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:

Layer	What it covers	Options
Softphone UI	The visible phone — buddies, streams, dial pad, call controls, settings	Browser-Phone (or build your own)
SIP signalling	The protocol layer: SIP over WebSocket	JsSIP, SIP.js
WebRTC media	Audio and video transport	Built 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 matters for developers who want to use it as a starting point rather than a finished product. You can brand it, restyle it, embed it in an iframe inside another application, or pull pieces apart and reassemble them to fit a specific product. Browser-Phone is a working phone you can ship, and a codebase you can shape.

The comparison everyone gets wrong

Most write-ups stack JsSIP and SIP.js side by side as if those were a developer’s only two options, and quietly assume the UI work is “trivial” or “outside scope.” The UI work is neither. Building a usable softphone UI from a blank page is genuinely months of work, and it’s the part most developers underestimate when they pick a signalling library and start coding.

If you take JsSIP or SIP.js alone, here is what you’ve signed up to build:

• A softphone UI from scratch — buddy list, dial pad, settings, presence, call history, notifications, the lot
• Call control logic — answer, hold, mute, transfer (blind and attended), conferencing
• Call recording on the client side, if you need it
• A messaging or notes layer if you want anything beyond calls
• Visual states for ringing, connecting, in-call, on hold, missed, declined
• Permissions flows for microphone and camera access, with the browser-specific quirks each one has

If you take Browser-Phone, all of that is in place. You can deploy it as-is, configure it for your PBX, and have a working phone in a single afternoon. You’re not skipping any signalling complexity — that’s still SIP, still WebSocket, still the same problems — but you’re skipping the months of UI work that JsSIP and SIP.js leave to you.

This is what makes the comparison feel uneven at first glance. JsSIP and SIP.js are libraries. Browser-Phone is a phone. They are not the same kind of thing. The reason to compare them anyway is that they answer the same underlying question — “how do I add SIP calling to a browser?” — at different levels of abstraction.

When JsSIP is the right answer

You want JsSIP when:

• You need granular control over the SIP layer — custom headers, unusual call flows, integration with a SIP server that has its own quirks
• You already understand SIP well, and the lower abstraction is an advantage rather than a hurdle
• You’re working with a SIP server like Kamailio or OpenSIPS where JsSIP has a particularly clean track record
• You’re building something so specific that no existing UI fits, and you have the engineering capacity to build one

JsSIP rewards developers who treat it as a kit. It’s reliable, mature, and has been quietly powering production WebRTC apps for over a decade.

When SIP.js is the right answer

You want SIP.js when:

• You want a higher-level signalling library with cleaner session management and better documentation
• You’re starting from scratch on the signalling layer and don’t want to learn SIP’s edge cases just to make outbound calls work
• You value an active maintainer with a commercial interest in keeping the library current
• You still intend to build the UI yourself, because none of the existing softphone projects fit your needs

SIP.js is the more accessible of the two signalling libraries, and for most developers building their own custom WebRTC client today, it’s the default sensible choice between the two.

When Browser-Phone is the right answer

You want Browser-Phone when:

• You want a softphone that already works, configured for your PBX, with a UI you don’t have to build
• You’re comfortable with AGPL-3.0 licensing and the obligations that come with it
• You like the buddy-and-stream model and want a softphone that organises information around the people you talk to rather than the calls themselves
• You want to brand, restyle or embed a softphone UI into your own product rather than designing one from scratch
• Your use case is a working phone for a working PBX — Asterisk, FreeSWITCH, FreePBX, FusionPBX, VitalPBX or 3CX — and you’d rather start from a complete project than a blank page

Browser-Phone is the right answer for the large majority of developers who arrive at this comparison without realising they had an option beyond the signalling libraries. It’s also a particularly common choice for sysadmins and Asterisk specialists who already operate their own PBX infrastructure and just want a usable phone to put in front of it.

The buddy-and-stream model: why it matters

Worth spending a moment on this because it’s the conceptual heart of what Browser-Phone is doing differently.

Traditional softphones — the kind that have been around since the early 2000s — are built around the call as the primary object. The screen shows you a list of recent calls, a list of missed calls, a dial pad, and somewhere off to the side, a contacts list. You make a call by finding a number; you receive a call from a number. Everything is organised around the call event.

The buddy-and-stream model inverts this. The primary object is the buddy — the person, the team member, the contact, the customer. Each buddy has a stream attached to them: a single timeline that contains the call history with that buddy, messages exchanged, recordings, notes, and any other information that can be associated with the conversation. When you bring up a buddy, you don’t bring up a phone number. You bring up the entire relationship.

For business communication, this is the right model. The thing that matters about a sales call last Tuesday isn’t that it happened at 14:32 for 23 minutes; it’s that it was the third conversation with that prospect this month, and the previous two are right there in the same view. The thing that matters about an internal call with a colleague isn’t the call log entry; it’s the ongoing thread of work that the call was part of.

The buddy-and-stream model also makes the UI extensible in a clean way. A stream can hold anything that’s logically tied to that buddy — call details, messages, files, custom integrations, CRM data, notes. The data model and the UI model agree.

This is the gap that JsSIP and SIP.js, as signalling libraries, leave entirely unfilled. They give you the ability to make and receive calls. They give you nothing about how to organise the information those calls produce. Browser-Phone fills that gap.

Compatibility with your existing PBX

All three options work with the major SIP servers — Asterisk, FreeSWITCH, Kamailio, OpenSIPS, 3CX, FreePBX, FusionPBX, VitalPBX — but the work involved differs.

With JsSIP or SIP.js, your PBX needs a WebSocket transport configured. Asterisk and FreeSWITCH support this, but the configuration isn’t trivial, and the codec negotiation between the browser and the PBX is your problem to solve. You’ll also be writing the UI layer that sits on top of the signalling.

With Browser-Phone, the PBX still needs a WebSocket transport, but the UI work is done and the configuration is documented for the major supported PBX platforms.

For teams with an existing PBX they don’t want to reconfigure heavily, this is often the deciding factor — Browser-Phone is the option that’s been tested in the wild against the most PBX configurations.

A short comparison table

	JsSIP	SIP.js	Browser-Phone
What it is	SIP signalling library	SIP signalling library	Complete open-source softphone
UI included?	No	No	Yes (buddy-and-stream)
You build	UI + everything around it	UI + everything around it	Mainly configuration and customisation
License	MIT	MIT	AGPL-3.0
Best for	SIP experts wanting low-level control	Developers wanting clean signalling APIs	Anyone who wants a working softphone
Cost	Free	Free	Free
Time to working call	Weeks-months	Weeks	Hours

A decision framework

If you’re trying to decide between the three, the questions to ask in order are:

1. Do I want to build a custom softphone UI, or use an existing one?

If you want to build your own — for reasons of design control, product fit, or sheer preference — drop to question 2. If you’d rather use a working softphone and customise from there, Browser-Phone is the natural starting point. Stop here.

2. How deep is my SIP expertise?

If you have engineers who know SIP well and want low-level control, JsSIP is a strong fit. If your team is competent but not SIP-specialist, SIP.js will be more productive.

3. How important is documentation and onboarding speed?

If “important,” SIP.js. If “the schedule isn’t the constraint, getting it exactly right is,” JsSIP is worth the deeper learning curve.

The category mistake to avoid is comparing the signalling SDKs in isolation and concluding that the cheapest sticker price wins. They’re all free. The cost that matters is the cost of getting from “library installed” to “working phone in production.” That cost is highest for JsSIP, lower for SIP.js, and lowest for Browser-Phone.

The takeaway

JsSIP and SIP.js are both good SIP signalling libraries. Both have powered real products for years. Between them, SIP.js is the more accessible choice today, while JsSIP rewards developers who want lower-level control and who already know SIP.

Browser-Phone is in a different category — it’s a complete, ready-to-deploy softphone with a configurable UI built around the buddy-and-stream paradigm. For developers who want a working phone rather than a starting point, that’s the more honest comparison.

The right choice depends on what you actually want to build. If you want to design every part of the phone yourself, pick a signalling library and start with the protocol. If you want a softphone that already works and you can shape from there, Browser-Phone is the answer.

Browser-Phone is also available as a managed service. If you want the same softphone, the same buddy-and-stream UI, and the same PBX compatibility — but without the work of self-hosting and operating the supporting infrastructure — Siperb is the commercial offering built on Browser-Phone. Version 4.0 of the Browser-Phone project is being developed under Siperb.