Documentation

General

MicroSIP offers two softphone modes: Single Call Mode (the default single-window interface) and Call Manager (two windows, supporting multiple calls, conferences, and attended transfers).

Communication can go through a SIP server or PBX, configured with Add Account, or as direct IP/domain calls through the Local Account, which needs no extra setup. The app minimizes to the system tray and requires no extra runtimes.

Dialpad

Used for dialing and sending DTMF tones. It accepts phone numbers, extensions, SIP addresses, and IP addresses. Advanced calls can use a full SIP URI with custom headers and an optional DTMF sequence after a comma.

Control switches: DND, FWD, AC (auto-add to conference), AA (auto-answer).
Buttons: CONF, REC.

Contacts

Only the Number field is required, and it must be unique.
Presence Subscription enables availability status, BLF, and call pickup, which may need PBX configuration. Asterisk is the exception for Publish Presence.
Call pickup uses the "**" prefix by default and is configured through Feature Codes.

Call Manager

Tab-based, with one tab per phone number. Switching tabs puts the current call on hold.
Supports conferences (one multi-party conference at a time), blind and attended transfers, instant messaging, and a per-tab activity log.
The Last Call button toggles between recent conversations. Closing a tab ends its call.

Account

SIP account: registers with a server or PBX; only one can be active at a time. Details come from your provider or administrator.

Local Account: direct calls without registration; disabled by default, and only one can exist.

Account settings include server, proxy and identity (username, domain, login, password, display name); dialing prefix and dial plan; caller ID hiding and voicemail number; media encryption (disabled, optional/mandatory SRTP, DTLS-SRTP); transport (UDP as failsafe, TLS as best; TCP may break behind SIP ALG); and public address, local port, publish presence, ICE, allow IP rewrite, and session timers.

Settings

Single call mode, ringtone, microphone amplification, software level adjustment.
Audio codecs, VAD, echo cancellation, forced incoming codec, H.264/H.263 toggles, video bitrate.
DTMF method (Auto, RFC2833, or in-band), auto answer, call forwarding, feature codes, deny incoming.
Directory of users: contacts or presence loaded over HTTP(s) in JSON or XML (Cisco, Yealink formats supported).
STUN server (not for symmetric NAT), media button and headset support, sound events, local account, log file, crash reports.

Advanced settings (microsip.ini only)

Command hooks: cmdCallStart, cmdCallEnd, cmdIncomingCall, cmdCallAnswer.
autoHangUpTime, maxConcurrentCalls, noResize, userAgent, multiMonitor.
Port knocker: portKnockerHost and portKnockerPorts.

DTMF, Video and Command line

DTMF: press dialpad keys during a call, or add a sequence after a comma in the number; each comma pauses about one second.

Video: H.264 (default) and H.263+ codecs at 640x480 and 30 fps. Uses DirectX/Direct3D and does not work over Remote Desktop.

Command line: options include dialing a number, hanging up, answering, transferring, sending DTMF, starting minimized, resetting configuration, and exiting.

Hotkeys: F2 answers a call and F4 ends all calls, when the main window is active.

Remark

SRTP, ICE, many codecs, or video can make SDP packets exceed the 1500-byte MTU and cause fragmentation, which some routers handle poorly. TCP or TLS transport can help.