MicroSIP is a lightweight open-source SIP softphone for desktop voice and video calls, built on the PJSIP stack and compatible with standard PBX systems and VoIP providers. It suits offices and support desks that want dependable calling, presence, and messaging without heavy software.
MicroSIP is a compact SIP softphone that turns a desktop into a working phone extension, registering with an IP-PBX, hosted provider, or any standards-compliant SIP server. Setup starts with an account: you enter the server address, extension, and password, then dial numbers, SIP addresses, or IP addresses straight from the dialpad. A quality indicator sits above the buttons, switching to the Call Manager opens tabs for simultaneous calls, conferences, transfers, and instant messages, and contacts show live presence so you know who is reachable. Everything lives in one small window that minimizes to the system tray and can start automatically. No extra runtimes or frameworks are required, and configuration sits in a single portable file that copies between machines.
Two details set MicroSIP apart in a crowded softphone market. First, it is portable: the whole configuration lives in an INI file next to the app, so an administrator can deploy identical settings by copying a file, and command-line switches let scripts answer, hang up, transfer, or dial without touching the interface. Second, it keeps advanced call control within reach, offering a Call Manager window for simultaneous calls, conferences, and attended transfers instead of a stripped-down dialer. Server-free calling is supported too, so two endpoints on the same network can talk by IP address with no registration. Teams that prefer a broader multiplatform client with conferencing features, such as Linphone, compare both before standardizing.
The clearest benefit of MicroSIP is that it stays out of the way. The dialer opens instantly, sits in the tray, and needs no additional runtimes, so an older office machine handles calls without the usual update churn. Hardware support is practical rather than token: media keys answer, hold, and hang up, HID headsets from major vendors drive call control and LED indication without vendor software, and the answer window can even follow whichever monitor you are using. Security covers what business buyers ask about, with TLS for signaling and SRTP or DTLS-SRTP options for media, plus dial plans and deny rules that block unwanted callers before the phone rings. Administrators get call recording, auto answer with delays and caller filters, call forwarding, feature codes for transfers, and command-line switches for scripted workflows. Contacts support presence, busy lamp field, and call pickup, and the interface works with screen readers and right-to-left languages. For teams that mainly need reliable voice, messaging, and presence on the desktop, MicroSIP delivers those with a fraction of the setup burden.
Key capabilities and highlights.
Calls run over a broad codec set that includes Opus, G.711, G.722, G.729, GSM, iLBC, Speex, and SILK, with WebRTC-based echo cancellation and voice activity detection to keep conversations clear. Video uses H.264 or H.263+ with adjustable bitrate caps, and an on-screen indicator reports live call quality.
Disabling single-call mode opens a tabbed Call Manager where each number gets its own tab holding a call, a message session, or both, and moving between tabs holds the previous call automatically. One multi-party conference and both blind and attended transfers are handled in this window.
Signaling can run over TLS, and media protection is selectable: optional SRTP, mandatory SRTP, DTLS-SRTP with an SRTP fallback, or DTLS-SRTP only. The documentation is refreshingly direct that the PBX still sees decrypted traffic, which helps administrators pick a transport policy matching their threat model.
Several SIP profiles can be configured and switched between, each carrying its own dial plan, dialing prefix, proxy chain, caller ID hiding, and voicemail number. An optional local account places calls to an IP address or domain with no registration at all, which is handy for internal testing or peer-to-peer links.
SIMPLE messaging sends one-to-one text over the SIP channel and appears in the same tab as the related call or number, while presence subscriptions color contacts by availability so you can see who is free. Publishing presence also enables busy lamp field monitoring and call pickup on PBXs that support those options.
A dedicated record button captures calls, while auto answer can fire after a delay or only for listed numbers and SIP headers, which suits help desks and intercom setups. Do not disturb, call forwarding, deny rules for unrecognized domains, and feature-code transfers round out everyday call routing.
Contacts store numbers in almost any format, grouped and sorted locally, and can subscribe to presence for live status and busy lamp field indicators. For larger organizations, the client pulls a shared directory over HTTP from JSON, XML, or common IP phone directory formats, refreshing it on a schedule.
Settings live in a portable INI file that can be copied between machines, and command-line switches dial numbers, answer, transfer, send DTMF, or hang up without opening the interface. Command hooks fire on call events, media keys and HID headsets drive call control, and a port knocker can satisfy firewall rules.
Browse historical releases and previous builds.
Common questions about MicroSIP.
MicroSIP is a SIP softphone, so people use it to place and receive voice and video calls, exchange messages, and manage presence through a SIP provider or office PBX. It registers as an extension, which makes it a practical desk phone replacement on a computer.
Yes. The project is open source and available at no cost, and its maintainer accepts donations to fund continued development. Businesses that need extra reassurance can lean on the published documentation and community channels rather than a paid support contract.
Yes, it places video calls using H.264 or H.263+ with a configurable bitrate ceiling, and higher bitrates are needed for fast-moving scenes or the picture degrades. Audio-only use is perfectly fine for contact centers that never need a camera feed.
It follows standard SIP, so it registers with Asterisk-based PBXs, FreePBX and 3CX deployments, hosted voice providers, and private SIP servers. Any implementation that honors the protocol should work, and account details come from your provider or PBX administrator.
Yes. An optional local account lets two endpoints call each other directly by IP address or domain name with no registration, which is useful for testing a link or running a small peer-to-peer setup. The local account stays available even when no SIP profile is configured.
Media encryption protects the hop between the softphone and the PBX, so keeping audio and video confidential in transit means also using TLS transport for signaling. The server itself still has access to the unencrypted stream by design, a limitation MicroSIP's documentation states plainly.
Yes. Because the entire configuration is stored in a portable INI file, an administrator can prepare one setup and copy it to each workstation, then control start-up behavior with command-line switches. Shared directory URLs and consistent account templates cut the per-user work considerably.
It handles HID headsets and multimedia key events, so a USB or Bluetooth headset button can answer, hold, resume, or end calls while LED indicators follow call state. The developer advises against installing the headset vendor's own software, since it can interfere with that standard interface.
MicroSIP is a lightweight open-source SIP softphone for desktop voice and video calls, built on the PJSIP stack and compatible with standard PBX systems and VoIP providers. It suits offices and support desks that want dependable calling, presence, and messaging without heavy software.
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