At some point in almost every WebRTC SIP project, you hit the same decision point: SIP.js or JsSIP? They are the two libraries that dominate this space, they have been around long enough to be battle-tested, and they are different enough that the choice actually matters. Picking the wrong one does not sink a project, but it can create friction you did not budget for.
This is a genuine comparison, not a sales pitch for either. We will look at where they come from, how their APIs differ, what each one handles better, and at the end, why Web Phone is built on SIP.js — with the actual reasons, not the marketing version.
First, some history — because it explains a lot
JsSIP came first. It was created by the team at OverSIP, including Iñaki Baz Castillo — who also co-authored RFC 7118, the spec that defines WebSocket as a transport for SIP. That is not a trivial credential. JsSIP was built with SIP standards compliance as a core principle, and it shows. It is precise, it is lean, and it has been actively maintained since the early days of WebRTC.
SIP.js is a fork of JsSIP. The OnSIP team created it while building their own WebRTC products and found that JsSIP, solid as it was, did not cover everything they needed — particularly around call forking, DTLS key exchange timing with legacy devices, and API structure. So they forked it, made substantial changes, and released it as SIP.js. The fork happened openly and in the spirit of open source collaboration rather than frustration.
Understanding that SIP.js started as a JsSIP fork matters because it tells you something important: these libraries share the same foundations. The differences are in what was built on top of them.
How the APIs compare
JsSIP keeps it compact
JsSIP’s core abstraction is the UA — the User Agent. You configure it, instantiate it, and call methods on it directly. For a project where you want to get a working softphone up quickly and you know your way around SIP, this feels natural. The API surface is deliberately small and the concepts map closely to SIP terminology.
The trade-off is that the monolithic UA design gets in the way once your application needs to manage multiple concurrent sessions, complex call state, or fine-grained control over the media pipeline. You can work around it, but you are working around it.
SIP.js separates its concerns
SIP.js breaks things into distinct objects — UserAgent, Registerer, Session — each responsible for its own slice of the call lifecycle. This is more code to write upfront, and the learning curve is slightly steeper. But for any application of meaningful complexity, that separation pays back quickly. You are not fighting the library when you need to do something non-trivial.
The TypeScript support in SIP.js is also considerably more complete. In 2026, if you are building anything serious in JavaScript, you are probably building it in TypeScript, and the difference in type coverage is noticeable.
Where each one has the edge
Call forking — SIP.js
When a single INVITE goes to multiple registered endpoints and the first to answer wins, SIP.js handles this more cleanly. It was one of the original motivations for the fork and the implementation is solid. If your deployment has users registered across multiple devices simultaneously, this matters.
Legacy PBX compatibility — SIP.js
Connecting a WebRTC browser client to a legacy SIP PBX involves a fair amount of negotiation — DTLS key exchanges, ICE candidates, codec agreement. SIP.js has had more work done in this area to reduce the edge cases: ghost calls, post-dial delays, negotiation failures. In mixed WebRTC/legacy environments, it tends to be more reliable in practice.
Bundle size — JsSIP
JsSIP is meaningfully smaller. If you are building a lightweight embedded widget where payload size matters, that is a real consideration. SIP.js is tree-shakeable in modern bundlers, so the gap narrows in practice, but JsSIP starts smaller.
Speed to prototype — JsSIP
If you know SIP well and want something working quickly, JsSIP’s compact API gets you there faster. Fewer abstractions to set up before you can make a call.
npm downloads (current)
SIP.js sits around 40,000 weekly downloads; JsSIP around 24,000. Both are healthy numbers for specialised libraries. SIP.js’s higher count likely reflects adoption in commercial contact centre and CRM platforms where TypeScript and concurrent session management are standard requirements.
Why Web Phone is built on SIP.js
We evaluated both properly before making a decision, and the choice came down to three specific requirements rather than general preference.
Web Phone is designed to embed into existing applications — CRMs, dashboards, helpdesk portals — where it needs to handle multiple concurrent sessions, complex transfer flows, and integration with external media pipelines. SIP.js’s separated architecture handles that more cleanly than JsSIP’s UA-centric model.
We also needed reliable behaviour in mixed WebRTC and legacy PBX environments. A significant proportion of the infrastructure Web Phone connects to is Asterisk or FreeSWITCH with varying degrees of WebRTC readiness. SIP.js’s DTLS handling and call forking support reduces the edge cases in those environments.
And the TypeScript coverage was genuinely important. Web Phone’s SDK is TypeScript-first, and SIP.js’s type definitions are complete enough to make the integration layer clean rather than something you have to paper over with type assertions.
JsSIP is a well-built library maintained by people who clearly know SIP inside out. For a different set of requirements — a lean embedded widget, a quick prototype, a project where JsSIP’s API just fits the mental model better — it is absolutely the right choice. For what Web Phone needs to do, SIP.js was the better fit.
A quick note on alternatives
SIP.js and JsSIP genuinely dominate this space in 2026. SIPml5 exists but has seen limited active development in recent years. For any production WebRTC SIP project, the decision is between these two — and whichever you choose, you are in good company.
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