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
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@example.com, firstname.lastname@example.org:5043, email@example.com, sip:192.168.1.55,
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
Allows you to make calls and exchange with instant messages with remote
party. To close tab page right click on tab.
- SIP server
Your account SIP server.
- SIP proxy
Your account SIP proxy.
Your account username.
Your account domain.
Username for authentication. If empty, will be used Username.
Your account password.
- Display name
Your name, remote party will see it in incoming calls and messages.
- Media encryption (remark 1)
Disabled - never use encryption, Optional - use encryption when remote
party supports encryption, Mandatory - use encryption always. Recommend
Depends from your SIP server configuration. Try one by one from TLS,
TCP, UDP. Recommend value: TLS.
- Public address
You can specify any address or hostname for this field, for example it can point to one of the interface address in the system. Only for local account: it can point to the public address of a NAT router where port mappings have been configured for the application. If you use microsip with SIP server, you can not change this setting to public address of a NAT router - it will not take effect.
- 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 contacts can see
your status. 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.
- 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.
- ICE (remark
Helps to find shortest way for media streams. It is usefull when
posible direct P2P connection without SIP provider mediagate. Against
ICE standard, in MicroSIP removed ICE mismatch check - this make
possible direct P2P connections between softphones if SIP server
changes IP address in "c=IN IP4 x.x.x.x" record of SDP. Recommended
value - enabled.
- Ringing sound
You can choose any WAV file on incoming call.
- 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
- Disable VAD
Disables voice activity detection. Default value - no.
- 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
- Auto answer
All incoming calls will be auto answered. MicroSIP will beep and popup
when call is accepted.
- Sound events
Playback key presses and signals of outgoing call.
- Disable local account
Local account allow you to receive incoming calls without SIP server
and account. In this case remote party must call you by your IP
address. Example: sip:192.168.1.21
- Single call mode
Disables message window when making calls. In this mode it looks
simplier, but you loose some functionality.
Settings not included in Settings dialog
You need to modify microsip.ini manually.
- "CmdIncomingCall" - runs specified command on incoming call and
pass caller ID as parameter.
While you are in call you can press buttons on dialpad to send DTMF
signals. It will be send to contact, defined by active tab in messages
dialog. DTMF digits sends as RFC 2833 events, if supported by remote party. If not - as in-band DTMF.
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
Video capture and video rendering uses DirectX and Direct3D (with
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
Call number: microsip.exe number
Start minimized: microsip.exe /minimized
- Remark 1
This feature increases SDP message length of INVITE query. Some SIP
servers, like Ekiga, works only via UDP protocol and does not support
fragmentation of big UDP packets. Enabling this feature with such SIP
servers can be reason of calls impossibility.