{"id":30,"date":"2026-04-07T16:23:09","date_gmt":"2026-04-07T16:23:09","guid":{"rendered":"http:\/\/localhost:8080\/knowledge-base\/browser-phone\/"},"modified":"2026-04-18T13:26:23","modified_gmt":"2026-04-18T13:26:23","slug":"browser-phone","status":"publish","type":"post","link":"https:\/\/www.siperb.com\/kb\/browser-phone\/","title":{"rendered":"Browser Phone"},"content":{"rendered":"\n

Browser Phone<\/strong> is an open-source JavaScript SIP\/WebRTC softphone \u2014 a fully functional telephone that runs in any web browser, connecting to your PBX without plugins, downloads, or proprietary hardware. It is the project that Siperb was built from, and it remains the core of Siperb’s Web Phone SDK<\/a> today.<\/p>\n\n\n\n

The Open-Source Project<\/h2>\n\n\n\n

Browser Phone was created to solve a specific and practical problem: making and receiving SIP calls directly from a web browser, using WebRTC, without any additional software. At the time, most softphone solutions required either a native app, a browser plugin (Flash, Java applets), or complex PBX-side WebRTC configuration. Browser Phone changed that by handling SIP signalling over a WebSocket connection and media via the browser’s built-in WebRTC stack.<\/p>\n\n\n\n

The original project is available on GitHub under the InnovateAsterisk\/Browser-Phone<\/a> repository. It remains free and open source \u2014 you can fork it, self-host it, and modify it with no licence cost and no Siperb account required.<\/p>\n\n\n\n

\"Browser<\/figure>\n\n\n\n

The project became widely used in the Asterisk and FreeSWITCH communities because it offered a ready-made WebRTC phone client that just worked. Developers could point it at their PBX and have a browser-based softphone running in an afternoon. That simplicity \u2014 and the permissive open-source licence \u2014 is what made it a starting point for so many VoIP projects, including Siperb.<\/p>\n\n\n\n

From Browser Phone to the Siperb Web Phone SDK<\/h2>\n\n\n\n

As self-hosted Browser Phone deployments grew, the limitations became clear. For most organisations, the real blocker was the requirement that the PBX itself support WebSocket connections and WebRTC media. Older Asterisk and FreeSWITCH installations \u2014 and almost all other PBX vendors \u2014 did not. This meant that to use Browser Phone, you either had to upgrade your PBX or run a WebRTC gateway alongside it.<\/p>\n\n\n\n

Siperb was created to remove that requirement. Starting from the Browser Phone codebase, we built the cloud infrastructure layer that sits between the browser and any PBX \u2014 handling the translation, the security, and the scalability that the self-hosted project could not provide out of the box.<\/p>\n\n\n\n

The Browser Phone JavaScript code remains the core of what we call the Browser Phone Core<\/strong> \u2014 the client-side engine that powers the Siperb Web Phone SDK. When you embed a Siperb-connected phone widget in your website, you’re running a hardened, cloud-connected evolution of the same open-source code.<\/p>\n\n\n\n

\"Siperb
Siperb as a Browser Tab \u2014 the Browser Phone Core running in production<\/figcaption><\/figure>\n\n\n\n

What Siperb Added<\/h2>\n\n\n\n

The gap between “open-source phone in a browser” and “production-ready softphone platform” is significant. These are the key pieces Siperb built on top of the Browser Phone Core:<\/p>\n\n\n\n