# Web Phone: WebRTC SIP phone for your website or CRM

> Web Phone is truly the “Swiss Army knife” of embedded Web Phones\. Whether you are using your own Web Phone SDK, SIP\.js, JsSIP or our free Browser Phone SDK, we have a solution for you\.

Generated by Yoast SEO v27.0, this is an llms.txt file, meant for consumption by LLMs.

## Pages
- [Documentation](https://www.web-phone.org/documentation/)
- [Resources](https://www.web-phone.org/resources/)
- [Click To Dial](https://www.web-phone.org/click-to-dial/)
- [Support](https://www.web-phone.org/support/)
- [FAQ](https://www.web-phone.org/faq/)

## Posts
- [Why your embedded web phone should never store SIP credentials in the browser](https://www.web-phone.org/2026/05/04/sip-credentials-browser-security/): SIP credentials don't belong in localStorage\. Here's what they actually represent, why the browser is a poor place to store them, and the short\-lived token pattern that does the job properly\.
- [SIP\.js vs JsSIP: comparing the two leading WebRTC SIP libraries](https://www.web-phone.org/2026/03/26/sipjs-vs-jssip-webrtc-sip-libraries-comparison/): SIP\.js and JsSIP are the two dominant WebRTC SIP libraries\. Here's an honest comparison of both — and why Web Phone is built on SIP\.js\.
- [Click\-to\-Call Is Moving Into CRMs in 2026 — Here’s Why](https://www.web-phone.org/2026/01/27/click-to-call-crm-trend-2026/): Click\-to\-call \(also commonly called click\-to\-dial\) is quickly becoming one of the most requested features in modern customer platforms—not as a “nice\-to\-have” widget, but as a core workflow tool for call centres, sales teams, and customer service operations\.
- [AI and Web Calling: Why Engineering Judgement Wins](https://www.web-phone.org/2026/01/22/ai-web-calling-engineering-judgement/): Conrad De Wet explains how AI is reshaping web calling—shifting the moat from “working code” to engineering judgement, reliability, security, and ownership\.
- [WebRTC: Media, Signaling, ICE, and Why WebRTC is so powerful](https://www.web-phone.org/2025/12/21/webrtc-media-signaling-ice/): WebRTC is often treated as a black box\. In this deep dive, we open it up — examining how signaling, NAT traversal, media pipelines, and security layers work together to deliver real\-time communication in the browser\. Expect code, protocol details, and hard\-won lessons from building real WebRTC systems\.

## Categories
- [WebRTC](https://www.web-phone.org/category/webrtc/)
- [News \& Announcements](https://www.web-phone.org/category/news-and-announcements/)
- [Resources](https://www.web-phone.org/category/resources/)
- [Security](https://www.web-phone.org/category/security/)

## Tags
- [SIP](https://www.web-phone.org/tag/sip/)
- [VoIP](https://www.web-phone.org/tag/voip/)
- [WebRTC](https://www.web-phone.org/tag/webrtc/)
- [Browser Phone](https://www.web-phone.org/tag/browser-phone/)
- [authentication](https://www.web-phone.org/tag/authentication/)

## Optional
- [Sitemap index](https://www.web-phone.org/sitemap_index.xml)