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,
firstname.lastname@example.org, email@example.com:5043, firstname.lastname@example.org, 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
"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.
- SIP server
Your account SIP server.
- SIP proxy
Your account SIP proxy. Examples: "192.168.1.1", "192.168.1.1:5070", "192.168.1.1;hide". ";hide" parameter can solve impossibility of registration or calls, when SIP server configured not the best way.
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.
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
You can specify IP address or hostname, it can point to one of the interface address,
it can point to the public address of a NAT router where port mappings have been configured.
If you use SIP server and have problems with network address, leave this in "Auto" and try "Allow IP rewrite" feature,
it will determine and update address automatically.
- 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.
- 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.
- Allow IP rewrite
Enable this only if you can not make calls without it. It can solve problems, connected with NAT configuration or multiple IPs. When this option is enabled, MicroSIP will keep track of the public IP address from the response of REGISTER request. This public IP will be used in later queries header and payload: Contact, VIA and SDP. Default: disabled.
Settings not included in Settings dialog
You need to modify microsip.ini manually.
- "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.
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
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.