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MicroSIP online help


MicroSIP can work with and without VoIP provider / SIP server. Second case uses more widely.
MicroSIP always starts minimized in tray. Right click on tray icon lets you access account and application settings.
MicroSIP not requires any preinstalled codecs, all codecs already included.


Used to enter numbers with mouse or sending DTMF signals. Also you can enter number with physical keyboard, in this case you can enter letters, specify custom domain and port. Examples: 13455674657, buddy, [email protected], [email protected]:5043, [email protected], sip:, etc.


To add contact right click on blank area. Contact number can be in any format, see examples in Dialpad.
If you enable "Presence subscription" MicroSIP will send on SIP server the subscribe query for contact presence. Your SIP server must support this feature.

"Presence subscription" allows to use BLF functionality - pickup incomings calls of other users. How BLF works in MicroSIP:

Users must enable "Publish presence" in Account window.
You must enable "Presence subscription" for users in your Contacts.
After that, users in your Contacts become colored. If they still grey - configure your SIP server (PBX).

When user from your Contacts will have incoming call, it will blink (green blinking icon). To accept incoming call, double click it or right click and use context menu - "Call pickup". Pickup code used in MicroSIP is "**" and currently cannot be changed.

User colors: green - online, red - offline, yellow - active call, green blinking - incoming call.


Allows you to make calls and exchange with instant messages with remote party. To close tab page right click on tab.


screenshot Account
  • SIP server
    Your account SIP server.
  • SIP proxy
    Your account SIP proxy or a chain of proxies. Examples: "", "", "", ";hide". ";hide" parameter can solve impossibility of registration or calls due to server configuration.
  • Username
    Your account username.
  • Domain
    Your account domain.
  • Login
    Username for authentication. If empty, will be used Username.
  • Password
    Your account password.
  • Display name
    Your name, remote party will see it in incoming calls and messages.
  • Dialing Prefix
    International calling prefix for numbers in local format (must begin with "+" or "00"); or a simple prefix for each dialing phone number.
  • Voicemail Number
    Voicemail access number. If empty, microsip will try to determine it automatically.
  • Media encryption (remark 1)
    Disabled - never use encryption, Optional - use encryption when remote party supports encryption, Mandatory - use encryption always. Recommend value: Optional.
  • Transport
    Depends from your SIP server configuration. Try one by one from TLS, TCP, UDP. Recommend value: TLS.
    Auto means mixing TCP and UDP. If your server do not support TCP, "Auto" will cause delay before outgoing call, switch to UDP.
  • Public address
    Can be used to solve call flow and media delivery issues when you don't have dedicated public IP address. You can manually specify IP address or hostname for Via, Contact and SDP. It can point to one of the interface address OR it can point to the public address of a NAT router where port mappings have been configured. For automatic public address detection and rewrite you can use Allow IP rewrite feature or use STUN server.
  • Local port
    By default MicroSIP tries to listen on standard SIP port - 5060. If port is busy by other application, MicroSIP will listen on random port. You can manualy change port to any.
  • Publish presence
    Sends on SIP server publish query, it means that other subscribed contacts can see your status and can pickup your incoming calls (BLF functionality). Besides, often you must specify which contacts have right to see your presence information - you can done this for example via SIP provider webpage. Your SIP server must support this feature.
  • ICE (remark 1)
    Helps to find shortest way for media streams and reduce media latency. It is usefull when posible direct P2P connection without SIP provider mediagate. Enabling ICE can cause problems with in media delivery if SIP server configured incorrecly.
  • Allow IP rewrite
    Can be used to solve call flow and media delivery issues when you don't have dedicated public IP address. If enabled, MicroSIP will keep track of the public IP address from the response of REGISTER request. Public IP will be used in later SIP queries in Via, Contact and SDP.
    See also: Public address, STUN.
  • Disable Session Timers
    Specify the usage of Session Timers. Try to disable Session Timers if your calls dropps after XX minutes. Recommended value: unchecked.


screenshot Settings
  • Ringtone
    You can choose any WAV file on incoming call.
  • Microphone Amplification
    Extends range of input signal level regulation by adding software amplification on top half of regulator. Default value - no.
  • Software Level Adjustment
    Enables internal input level regulation instead of changing global level of input device. Note that hardware regulation has lower noise rating. Default value - no.
  • Audio Codecs (remark 1)
    You can enable and disable codecs by moving it between lists. Also you can set codec priority (for outgoing calls) by moving codecs in right list.
  • VAD
    Enables voice activity detection. Default value - no.
  • EC
    Enables echo cancellation. Default value - yes.
  • Force codec for incoming
    Normally, caller defines codecs priority. For incoming calls this option allows you (callee) select prefered codec.
  • Disable H.264 codec
    Normally caller defines codec that will be used by both parties. But some callees parties forces your selected codec with some other, but in same time they supports your codec. In this case you can disable unwanted codec. Default value - no.
  • Disable H.263 codec
    See above. Default value - no.
  • Video codec bitrate
    Set the maximum bitrate. If one party set 256 kbit/s and other 512 kbit/s - will be used 256 kbit/s for both. Dynamic scenes requires higher bitrates (~512 kbit/s), otherwise picture quality will fall down.
  • DTMF Method
    Auto: MicroSIP will use RFC2833 for DTMF relay by default but will switch to in-band audio DTMF tones if the remote side does not indicate support of RFC2833 in SDP. Note: in-band method will not work properly with every audio codec due to compression algorithms.
  • Auto answer
    MicroSIP will play short tone and popup when call auto accepted. SIP header - when receiving the "Call-Info: Auto Answer" or "Call-Info: answer-after=0" or "X-AUTOANSWER: TRUE" in SIP header.
  • Deny incoming
    Helps to block unwanted or spam incoming calls. Different user/domain/user-domain means that callee data do not match data in your account window. Different remote domain means that caller domain do not match domain in your account window.
  • Directory of users
    Enter URL to obtain contacts from external source via HTTP(s). JSON and XML responses are supported.

    JSON format:

    {"number": "001", "name": "Name 1", "info": "Optional text info", "presence": 0},
    {"number": "002", "name": "Name 2", "presence": 1}

    XML format:

    <?xml version="1.0"?>
    <name>Name 1</name>
    <info>Optional text info</info>
    <name>Name 2</name>
    "name" and "presence" are optional. Also supported Cisco IP phone directory format CiscoIPPhoneDirectory, Yealink and some other.
    To specify refresh rate value use "Cache-Control: max-age=3600" header, where 3600 - value in seconds. Default refresh value 3600.
  • STUN server
    Helps to make direct way for media streams without SIP provider media gate when NAT used. It open UDP ports on NAT server for incoming connections. Exists different NAT types (full cone NAT, (address) restricted cone NAT, port restricted cone NAT and symmetric NAT). You can use STUN only if your NAT is not symmetric! Otherwise you will have problems - you can not hear and can not hears you - remove it from settings. Default value - empty.

    how works STUN
  • Handle Media Buttons
    Enables handling of media keys or buttons events on multimedia keyboards or headsets with buttons (WM_APPCOMMAND message). Can be used for call answer, hold, resume and end call.
  • Sound events
    Playback key presses and signals of outgoing call.
  • Enable local account
    Local account allows you make and receive calls without SIP server and SIP account. In this case you can call by IP address (or domain name) as number.
    Note: local account always enabled if SIP account is not configured or disabled.
    Example: sip: or just or [email protected]
  • Single call mode
    Activates simple user interface. You should disable this if you wish to manage multiple calls, make attended transfers or conference calls.
  • Enable log file
    Activates microsip log file. Used for debugging. To open log file right click on tray icon.
  • Random position of the answer box
    Display incoming call window at random position on the screen and random monitor if many.
  • Send crash report
    Automatically send crash report to the microsip team for analyse. Report includes OS name and version, log file (if enabled in Settings). It never contains your passwords.

Settings not included in Settings dialog

You need to modify microsip.ini manually.
  • "sourcePort=5060" - use static source port of outgoing SIP requests (UDP transport only).
  • "cmdCallStart" - runs specified command when connection established. Caller ID passed as parameter.
  • "cmdCallEnd" - runs specified command when call ended. Caller ID passed as parameter.
  • "cmdIncomingCall" - runs specified command when incoming call arrives. Caller ID passed as parameter.
  • "cmdCallAnswer" - runs specified command when user answers on incoming call. Caller ID passed as parameter.
  • "autoHangUpTime"
  • "maxConcurrentCalls"
  • "noResize"
Port knocker feature. Send sequential UDP requests to a specified ports on a specific host (SIP server by default) before microsip tries the SIP registration. That allows SIP server to whitelist cliend IP in the firewall.
  • "portKnockerHost=host.com" - domain name or IP address of knocking host. If empty and port list isn't empty - SIP server value will be used.
  • "portKnockerPorts=1111,2222" - one or more ports separated by comma. If empty - feature disabled.


While you are in call you can press buttons on dialpad to send DTMF signals. If you want automatically pass DTMF commands just after call established, then add ",dtmf_sequence" or ",dtmf_sequence1,dtmf_sequence2" in calling number. One comma means pause in one second.


Supported H.264 and H.263+ (other name H.263-1998) video codecs. Default codec - H.264, video format - 640x480 @ 30 fps, outgoing bitrate 512 kbit/s. H.264 encoding requires significant CPU resourse. Recommended dual core processor, multimedia extensions like MMX will be used if is present.
Video capture and video rendering uses DirectX and Direct3D (with hardware acceleration).
Because hardware acceleration is used, video calls will not work with remote desktop session (RDP).
If you have serious problems with performance:
- update video adapter drivers
- install/reinstall DirectX (can be downloaded here)

Command line

Call a number: microsip.exe number
Hang up all calls: microsip.exe /hangupall
Answer a call: microsip.exe /answer
Start minimized: microsip.exe /minimized
Exit: microsip.exe /exit


  • Remark 1
    This feature increases UDP packet size (SDP message length of INVITE query). If UDP packet size will be > 1500 bytes (MTU), it will be fragmented. Not all routers can correctly work with fragmented UDP packets. So, if you enable extra feature like SRTP, or ICE, or select too many enabled codecs, or make video call, be ready that you will not be able make a call. Best exits from situation - use TCP or TLS transport, but in this case your SIP server must support it.