JsSIP vs SIP.js vs Browser-Phone: WebRTC SIP Compared

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JsSIP vs SIP.js vs Browser-Phone: Choosing an Open-Source WebRTC SIP Stack in 2026

Justin

2 June 2026

Short answer: JsSIP and SIP.js are open-source JavaScript libraries that handle SIP signalling — they speak the protocol but don’t give you a phone interface. Browser-Phone is an open-source softphone built on top of SIP signalling, with a complete, configurable UI organised around a buddy-and-stream paradigm. All three are free. The real question isn’t which one is “best” — it’s whether you want just a signalling library, or a signalling library plus a usable phone.

If you’re a developer deciding how to add browser-based calling to a web app, a CRM integration, a softphone product or an internal tool, this is the comparison that matters. Most other “best WebRTC SIP library” articles compare JsSIP and SIP.js as if those were the only two options. They aren’t. Browser-Phone sits at a different layer — it includes a UI — and that single difference reframes the whole decision.

The honest framing: it’s about layers, not features

Before comparing them, it helps to see what each one actually does:

LayerWhat it coversOptionsSoftphone UI The visible phone — buddies, streams, dial pad, call controls, settingsBrowser-Phone (or build your own)SIP signalling The protocol layer: SIP over WebSocketJsSIP, SIP.jsWebRTC media Audio and video transportBuilt into the browser

JsSIP and SIP.js sit at the signalling layer. They handle the SIP conversation between the browser and the SIP server — registration, INVITE, BYE, the works. What they don’t do is draw a phone on the screen. There is no buddy list, no dial pad, no settings panel, no presence indicator, no call history. If you want any of that, you build it yourself.

Browser-Phone sits a layer above. It uses SIP signalling under the hood, but adds a complete softphone UI on top, plus call recording, transfer, conferencing, and the other things a real phone needs to do.

The honest question isn’t “which library has the cleanest API.” The honest question is “how much of the phone am I willing to build myself?”

What each of these actually is

JsSIP

JsSIP is a JavaScript SIP signalling library, first released in 2011 and maintained by a small team in Spain. It implements SIP over WebSocket — specifically RFC 7118 — and uses the browser’s native WebRTC stack for media. It is unopinionated, low-level, and gives you direct access to the SIP layer.

What JsSIP does not do is draw a softphone on the screen. There is no buddy list, no dial pad, no settings panel, no presence indicator, no call history. If you want any of that, you build it yourself.

The library is stable rather than fast-moving — major releases come years apart. SIP itself isn’t moving fast either, so that isn’t necessarily a problem, but browser-side WebRTC changes can occasionally outpace the library, and you’ll be the one patching around them.

SIP.js

SIP.js began as a fork of JsSIP and has since gone its own way. It’s backed by OnSIP, a hosted SIP service, and benefits from a more active release cycle and better documentation. Where JsSIP is low-level, SIP.js layers a higher-level API on top — a SimpleUser class, clearer session management, more help for developers who don’t want to learn SIP RFCs cover to cover.

In raw download numbers, SIP.js is the more popular choice, with roughly 44,000 weekly npm downloads against JsSIP’s 25,000. The community is wider, the surface area of examples is bigger, and the onboarding curve is shorter.

But like JsSIP, SIP.js is signalling only. There is no UI. The same blank page problem applies: you’ve got the protocol, you don’t have the phone.

Browser-Phone

Browser-Phone is an open-source softphone project, AGPL-3.0 licensed, currently at version 0.3.x and hosted under the InnovateAsterisk GitHub organisation. It is what JsSIP and SIP.js stop short of — a complete, ready-to-deploy softphone with the UI work already done. The project supports Asterisk, FreeSWITCH, FreePBX, FusionPBX, VitalPBX and 3CX out of the box.

The defining choice in Browser-Phone is its buddy-and-stream paradigm . Most softphones organise the interface around calls — recent calls, missed calls, dial pad, contacts. Browser-Phone organises it around buddies, where each buddy has an associated stream that contains everything related to that contact: call history, messages, recordings, notes, and anything else that can be attached to a conversation. The unit of organisation is the person, not the call. When you bring up a buddy, you bring up the entire relationship with that buddy in one view.

This sounds like a small thing. It isn’t. It’s the difference between a softphone that feels like a 2005 desk-phone replacement and one that feels like a modern communications tool. It’s also what Browser-Phone is known for in the WebRTC and Asterisk communities.

The UI is configurable in a multitude of ways, which...