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Version history for MicroSIP Lite

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Changes for v3.19.14 - v3.19.15

  • - auto unmute on a new call
  • - ALT to set focus on the menu button
  • - show blocked incoming calls on the call page
  • - show original failed call message
  • - fixed VP8 bitrate setting



Changes for v3.19.10 - v3.19.11

  • - settings are divided into 2 columns
  • - reject call with busy command (decline in the past)
  • - added ability to specify multiple STUN or NS
  • - hide voicemail button
  • - pjsip update 5931
  • - fixed call recording path
  • - other improvements



Changes for v3.19.8 - v3.19.10

  • - added call recording (all calls or manual recording)
  • - added contacts CSV import/export
  • - added command events: cmdOutgoingCall, cmdCallRing, cmdCallBusy
  • - added number column in contacts list
  • - fixed command events execution sequence
  • - fixed codepage in audio and video devices names
  • - other fixes



Changes for v3.19.7 - v3.19.8

  • - added DNS SRV option
  • - added support of UTF8 and UTF16 in Google CSV import
  • - added tooltips for bottom buttons
  • - added bugfix for CANCEL outgoing call
  • - updated VP8 codec



Changes for v3.19.5 - v3.19.7

  • - audio codecs adapted for screen readers, manage with Space and Delete keys
  • - changed Dialer resizing
  • - fixed Opus codec
  • - fixed several possible crashes
  • - pjsip update 5861



Changes for v3.19.3 - v3.19.5

  • - missed call tray icon
  • - AA and DND buttons changes
  • - changed crash report
  • - pjsip update 5851



Changes for v3.18.5 - v3.19.3

  • - missed call tray icon
  • - AA and DND buttons changes
  • - changed crash report
  • - pjsip update 5851
  • - new layout
  • - subscribe text info for contact (new column)
  • - on-the-phone presnece status
  • - changelog in update dialog
  • - system fixe



Changes for v3.18.3 - v3.18.5

  • - import contacts (Google CSV format)
  • - AMR codec parameters changed to be compatible with Android SIP dialer
  • - small fixes



Changes for v3.18.2 - v3.18.3

  • - added RTP port range setting
  • - added SIP source port setting
  • - added "rport" option
  • - added possibility to make call with media button
  • - fixed window focus at startup



Changes for v3.17.8 - v3.18.2

  • - improved compatibility in SDP negotiation
  • - increased maximum SIP packet length
  • 3.18.1 [MicroSIP-3.18.1.exe | portable] (1431 downloads), [MicroSIP-Lite-3.18.1.exe | portable] (296 downloads)
  • - high quality WebRTC echo canceler instead of Speex
  • - echo canceler enabled by default
  • - Public Address setting now affects also on Via and Contact headers
  • - rejecting an offered stream in SDP according RFC 3264
  • - minor fixes



Changes for v3.17.3 - v3.17.5

  • - UI changes
  • - fixed multi instance management in Wine



Changes for v3.16.9 - v3.17.3

  • - added option "Bring to Front on Incoming Call"
  • - project has moved in new IDE and built with modern compiler
  • - update library SDL 2.0.7
  • - update library VPX 1.7.0
  • - update library ffmpeg 3.4.1
  • - update library x264 0.152
  • - new G.729 codec implementation (annexes A and B supported)
  • - software video render for RDP connection
  • - other fixes



Changes for v3.16.7 - v3.16.9

  • - fixed media buttons
  • - fixed broken info in crash report
  • - settings dialog internal edits
  • - added AMR-WB codec
  • - added option: "Handle Media Buttons"
  • - improved multi instance management
  • - exploit protection (enabled SafeSEH, removed shared memory block)



Changes for v3.16.1 - v3.16.4

  • - added redial last number button
  • - added G.723 codec (no licence, limited usage)
  • - added possibility to pass DTMF automatically
  • - added auto answer after by timeout "Call-Info: answer-after=5"
  • - fixed application crash
  • - fixed H.264 video
  • - fixed save bitrate
  • - fixed make active option in menu
  • - code optimization
  • - update openssl 1.1.0f
  • - update pjsip 2.7.1



Changes for v3.15.10 - v3.16.1

  • - pjsip update 2.7
  • - openssl update 1.1.0f
  • - pass DTMF commands after call established (number,DTMFsequence1,DTMFsequence1,,,DTMFsequence3), one comma means pause in one second
  • - auto answer after timeout "Call-Info: answer-after=5"
  • - reverted make active option in menu
  • - small fixes



Changes for v3.15.9 - v3.15.10

  • - shortcuts feature (configurable buttons)



Changes for v3.15.7 - v3.15.9

  • - adjusting speakers volume only for calls
  • - HW/SW level microphone adjustment option
  • - microphone amplification option
  • - mute improvement
  • - improved VU-meters
  • - fixed VU-meters for WinXP
  • - fixed VU-meters for conference call
  • - fixed VU-meters for Wine (Linux)
  • - other fixes



Changes for v3.15.6 - v3.15.7

  • - VU meters and volume buttons
  • - added ini settings: autoHangUpTime, maxConcurrentCalls, noResize
  • - fixed hiding voicemail icon
  • - fixed call ending notification sound



Changes for v3.15.5 - v3.15.6

  • - voicemail feature
  • - disable session timers option
  • - fixed date/time display in Calls list



Changes for v3.15.4 - v3.15.5

  • - save position of incoming call window
  • - port knocker feature (UDP-ping SIP server before register, see help for details)
  • - added possibility to dial numbers with parameters and headers. Example: sip:user@host:port;param=pval?header=hval
  • - STUN server moved into Settings
  • - small fixes



Changes for v3.15.3 - v3.15.4

  • - bug fixes



Changes for v3.15.1 - v3.15.3

  • - DTMF method option (Auto, In-band, RCF2833, SIP-INFO)
  • - audible remote DTMF signals (RCF2833, SIP-INFO)
  • - mute microphone while sending in-band DTMF
  • - fixed bug in dialing from command line
  • 3.15.2
  • - pjsip update 2.6
  • - small edits



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