MicroSIP Setup: Connect the Softphone to SIP

MicroSIP is a free, open-source Windows softphone that registers directly to any standards-based SIP provider or PBX — Menu › Add Account, enter your SIP server, username, and password, and you are making calls. Built on the PJSIP stack, it is one of the fastest ways to put a carrier account on a desktop: no PBX in the middle, no licensing, a couple of megabytes of client.

This guide — from SIPNEX, an FCC-licensed carrier — covers the account fields that trip people up, the settings behind clean audio, and where a lone softphone stops being the right tool.

Adding an account

Open the main menu (top-right icon, or right-click the title bar) › Add Account:

• Account name: anything — it’s a label.
• SIP server: your provider’s server address.
• SIP proxy: only if your provider specifies one; otherwise leave empty.
• Username: the account’s user part (what forms username@domain).
• Domain: the SIP server hostname again for a typical username@sipserver account, or the domain part your provider specifies.
• Login: the authorization username — the field that causes most failed registrations. It silently defaults to Username when blank, so if your carrier issues a separate auth ID, it must go here.
• Password and a Display name for outbound caller presentation.
• Transport: UDP is the failsafe; TLS is the vendor-recommended option when your provider supports it. Media encryption (SRTP / DTLS-SRTP) is available but defaults to Disabled — turn it on per-account if your carrier supports encrypted media.

You can save multiple accounts, but only one is active at a time — MicroSIP is a single-line client by design.

Codecs and DTMF

Under Menu › Settings, codecs move between Available and Enabled lists; the order of the Enabled list is your outbound preference. For a US carrier account, put G.711u (PCMU) first — it’s the PSTN-quality baseline — with Opus a strong second for internet-quality calls between soft clients.

DTMF defaults to Auto, which negotiates RFC 2833 telephone-events when the far side supports them and falls back to in-band tones when it doesn’t — the correct behavior against carrier trunks. An explicit method selector is in Settings if a provider demands one mode.

NAT: the three options that matter

MicroSIP’s vendor guidance here is unusually specific:

• STUN: configure a STUN server only behind non-symmetric NAT — behind symmetric NAT, STUN actively causes audio-delivery problems.
• Allow IP rewrite: lets the client learn its public address from REGISTER responses — the low-effort fix for most home/office NAT.
• Public address: a manual override for the IP used in Via/Contact/SDP when you know your public address and want no guessing.

Also worth enabling on flaky providers: Disable Session Timers, which prevents mid-call drops on long calls when a far end mishandles session refresh.

Where MicroSIP fits — and where it doesn’t

A softphone registering straight to the carrier is perfect for a solo operator, a test endpoint while you bring up a new trunk, or a lightweight second line. What it is not is a phone system: one active account, no queues, no IVR, no shared lines. The moment you need extensions and routing, put a PBX in the middle — the same SIPNEX account that registers in MicroSIP plugs into any SIP-capable PBX, and MicroSIP then registers to the PBX as an extension instead.

We keep MicroSIP in our own toolbox for exactly one job: fastest possible validation that a trunk credential set registers and passes two-way audio before a PBX config gets involved.

Frequently asked questions

SIPNEX is an FCC-licensed carrier — the same account that registers in MicroSIP scales to a full PBX SIP trunk with unlimited channels and A-level STIR/SHAKEN attestation. Get SIP credentials or see rates.