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MicroSIP online help

General

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.

Dialpad

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, buddy@sip.com, buddy@sip.com:5043, buddy@192.168.1.43, sip:192.168.1.55, etc.

Contacts

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.

Calls/Messages

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

Account


  • 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 registration problem, when SIP server configured incorrectly.
  • 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.
  • 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.
  • 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 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 1)
    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


  • 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 list.
  • Disable VAD
    Disables voice activity detection. Default value - no.
  • Force codec
    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.
  • 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
    URL (with http://) of the file in XML format containing contacts list. Generic format:
    <?xml version="1.0"?>
    <directory>
    <entry>
    <name>Name</name>
    <extension>1234</extension>
    </entry>
    </directory>
    Also supported Cisco IP phone directory format CiscoIPPhoneDirectory and some other.
    To specify refresh rate value use "Cache-Control: max-age=3600" header, where 3600 - value in seconds. Default refresh value 3600.
  • 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.
  • Enable log file
    Activates microsip log file. Used for debugging. To open log file right click on tray icon.

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.

DTMF

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.

Video

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 number: microsip.exe number
Start minimized: microsip.exe /minimized

Remarks

  • 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.