head to head · open source
webrtc vs SIP.js
webrtc has 16,874 GitHub stars, 1,916 forks, 125 open issues and last shipped 2 days ago. SIP.js has 2,104 stars, 724 forks, 97 open issues and last shipped 5 months ago. webrtc leads on adoption by 702% (16,874 vs 2,104 stars). webrtc is written in Go under MIT; SIP.js is written in TypeScript under MIT. webrtc has attracted 11% as many forks as stars, SIP.js 34%. webrtc was the more recently maintained of the two, and both are self-hostable with no licence fee.
Two open source projects, one decision. Both are free and self-hostable — the differences are community size, license terms, language stack and release pace.
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Side by side
| webrtc | SIP.js | |
|---|---|---|
| GitHub stars | ★ 17K | ★ 2.1K |
| License | MIT | MIT |
| Written in | Go | TypeScript |
| Last push | 2026-11-02 | 2026-06-15 |
| Forks | ⑂ 1.9K | ⑂ 724 |
| Self-hosting | Yes | Yes |
| Data ownership | Your server | Your server |
pick webrtc if
- You weight community size — 17K stars and counting
- You want the MIT license terms
- Your stack matches Go
- You value the larger contributor base for long-term maintenance
pick SIP.js if
- You want the SIP.js feature set and don't need the biggest community
- You prefer the MIT license terms
- Your stack matches TypeScript
- You evaluated both and SIP.js fits your workflow better
About webrtc
Pion WebRTC A pure Go implementation of the WebRTC API New Release Pion WebRTC v4.0.0 has been released! See the release notes to learn about new features and breaking changes. If you aren't able to upgrade yet check the tags for the latest v3 release. We would love your feedback! Please create GitHub issues or Join the Discord to follow development and speak with the maintainers. Usage Go Modules are mandatory for using Pion WebRTC. So make sure you set export GO111MODULE=on , and explicitly specify /v4 (or an earlier version) when importing. example applications contains code samples of common things people…
read the full webrtc overview →
About SIP.js
SIP Library for JavaScript Create real time peer to peer audio and video sessions via WebRTC Utilize SIP in your web application via SIP over WebSocket Send instant messages and view presence Support early media, hold and transfers Send DTMF RFC 2833 or SIP INFO Share your screen or desktop Written in TypeScript Runs in all major web browsers Compatible with standards compliant servers including Asterisk and FreeSWITCH Demo Want see it in action? The project website, sipjs.com, has a live demo. Looking for code to get started with? This repository includes demonstrations which run in a web browser. Usage To…
read the full SIP.js overview →
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Frequently asked questions
Is webrtc or SIP.js more popular?
webrtc has 16,874 GitHub stars and SIP.js has 2,104. webrtc has the larger community by that measure.
Are webrtc and SIP.js free?
Both are open source. webrtc is licensed under MIT and SIP.js under MIT. Neither carries a licence fee.
What is the difference between webrtc and SIP.js?
webrtc is written in Go and SIP.js in TypeScript. The practical differences are community size, licence terms, language stack and release cadence — all compared in the table above.
Which should I choose, webrtc or SIP.js?
Choose webrtc if you want the larger community (16,874 stars) or its MIT licence terms. Choose SIP.js if its feature set, stack or MIT licence fits better. Both are self-hostable.