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
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,
firstname.lastname@example.org, email@example.com:5043, firstname.lastname@example.org,
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.
enable "Presence subscription" for users in your Contacts.
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
- 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 domain.
authentication. If empty, will be used Username.
- 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 value:
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
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
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
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
- 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
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:
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.
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 present.
Video capture and video rendering uses
DirectX and Direct3D (with hardware acceleration).
hardware acceleration is used, video calls will not work with remote
desktop session (RDP).
If you have serious problems with
- update video adapter drivers
install/reinstall DirectX (can be downloaded
Call a number: microsip.exe number
Hang up all calls:
Answer a call: microsip.exe /answer
Start minimized: microsip.exe /minimized
Exit: microsip.exe /exit
- Remark 1
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.