Changelog for Blink (Windows)

Version 1.4.2

September 11th, 2015

  • Fixed default URI after processing Google contact updates

Version 1.4.1

June 18th, 2015

  • Fixed opening files in binary mode

Version 1.4.0

June 10th, 2015

  • Added support for VP8 video codec
  • Added file transfer resuming capability
  • Added basic support for RTCP PLI feedback indication and use it to send / request keyframes
  • Fixed crash when handling bogus Opus packets

Version 1.2.1

March 24th, 2015

  • Added Video support
  • Added ZRTP support
  • Added support for file drag and drop on the chat window
  • Improve the performance of painting the video preview
  • Fix stopping composing timer when session is ended or chat is removed
  • Removed unused settings
  • Always play an audio alert when a file transfer finishes
  • Removed session and RTP timeouts from preferences
  • Reorganized preferences
  • Better icons for chat, logging and advanced settings
  • Updated test numbers
  • Check for the audio session item existance before operating on it
  • Ignore hold/hangup keyboard shortcuts if no session is selected
  • Render status messages appropriately
  • Optimized chat inline images to avoid compression artifact


July 30th, 2014

  • Fix version number


July 30th, 2014

  • Fix installer not removing old files from destination folder

Version 0.9.1

July 28th, 2014

  • Automatically transform URLs in the chat window into clickable links
  • Show P2P icon if contact is a Bonjour neighbour
  • Fixed handling unicode in user's home directory
  • Fixed http URL regular expression
  • Removed menu entry for buying prepaid credit
  • Fixed encoding of cut text for the VNC client
  • Rebuilt Blink and its dependencies using MinGW toolchain

Version 0.9.0

June 27th, 2014

  • Fix Blink not starting on Windows 7 and 8
  • Added screen sharing support
  • Refactored session info panel
  • UI Layout fixes
  • Added file transfer to the list of client capabilities
  • Fixed status message when audio is removed vs when the session is ended
  • Fixed exception when trying to change status after audio session is gone
  • The file transfer download directory cannot be undefined
  • Raised python-sipsimple version dependency

Version 0.8.2

May 28th, 2014

  • Fix corrupted incoming file transfers
  • Use a generic fail reason when none is provided
  • Add support for adding/removing multiple streams at once
  • Refactored SDP handling
  • Send re-INVITE after ICE negotiation is done
  • Enable RTP keep-alive using empty RTP packets
  • Disabled speex codec by default
  • Fixed race condition when saving ICE state
  • Fix closing media transport to avoid leaking STUN sockets
  • Use a shorter timeout for re-INVITEs that need to be answered without user interaction

Version 0.8.1

April 24th, 2014

  • Fixed issue with not fetching Google Contacts after recent Google changes
  • Changed audio codec order: prefer G722 over Speex

Version 0.8.0

April 11th, 2014

  • Added file transfer support using MSRP protocol
  • Added chat themes support
  • Added chat preferences
  • Added quick setting for playing message alerts
  • Added keyboard shortcuts for different windows
  • Added option for auto-accepting chat from known contacts
  • Rearranged menus in the main window
  • Refactored history menu
  • Improved displaying the account states in preferences and the main window
  • Save conference rooms in combobox history and limit it to 20 entries
  • Fixed unwanted selection switching when sessions were ended
  • Fixed using correct avatar and display name for received chat messages
  • Fixed updating toolbar buttons when changing sessions and allow hold early
  • Fixed using secondary ringtone when adding chat to an audio session
  • Fixed playing hold tone when switching sessions
  • Fixed playing tones when streams are cancelled
  • Fixed incoming dialog margins
  • Fixed processing DTMF tones
  • Close dialog if session is ended while in a proposal
  • Only remove the TLS certificate if it's in the application data directory
  • Play sound for received chat messages, if enabled
  • Do not play the hangup tone for sessions without audio
  • Do not close all windows when closing the main window

Version 0.7.0

March 5th, 2014

  • Added session information panel into the chat window
  • Added server side conferencing support
  • Automatically accept in-dialog chat proposals
  • Fixed potentially processing multiple settings changes at once
  • Enabled showing logs directory
  • Limit the time window for processing a DNS lookup to the current session
  • Tons of UI tweaks

Version 0.6.0

December 16th, 2013

  • Added chat sessions using MSRP protocol
  • Added Opus codec version 1.1
  • Fixed Opus codec negotiations bugs
  • Fixed unpickling for Bonjour Neighbours
  • Use single global c line when creating SDP
  • Fixed several memory leaks
  • Simplified processing Google contacts authorization and fixed some bugs
  • Set focus to the appropriate widgets during Google contacts authorization
  • Improved selection of the winning presence state
  • Removed donate menu action
  • Properly patch dnspython to make it non- blocking
  • Adjusted to the latest changes in SIP SIMPLE Client SDK

Version 0.5.0

August 9th, 2013

  • Adapted to changes in SIP SIMPLE Client SDK
  • Enabled Opus codec
  • Set default sample rate to 32 kHz
  • Fixed exception if Google contact has no name nor company
  • Fixed losing contact icons
  • Fixed computing hours in history entries
  • Fixed setting display name in history entry when URI is a phone number
  • Avoid publishing presence state twice when xcap settings change
  • Allow Ctrl+Delete/Backspace to hangup sessions because KDE steals Ctrl+Esc
  • Raise and activate preferences window when triggered if already visible
  • Do not allow toolbar to be hidden

Version 0.4.0

June 26th, 2013

  • Added support for multiple URIs per contact
  • Added support for Bonjour presence
  • Added minimize to system tray
  • Added text eliding using fading colors for labels with long texts
  • Adjusted selected group background colors for better visibility
  • Save and restore main window geometry across restarts
  • Move contact when its name changes to keep contacts sorted
  • Updated test contact icon files
  • Strip URI domain in history if URI looks like a phone number
  • Fixed creating offset-naive datetime for epoch
  • Fixed processing when initial winfo document is not full
  • Fixed publishing internal presence states
  • Fixed losing presence state when contact is modified

Version 0.3.1

April 25th, 2013

  • Fixed contact drag and drop
  • Fixed handling PIDFs from multiple accounts at once

Version 0.3.0

April 3rd, 2013

  • Added support for SIP PUBLISH method
  • Added subscriptions for presence (presence event package)
  • Added handler for outgoing presence (RLS SUBSCRIBE and RLMI NOTIFY)
  • Added handler for incoming presence (presence.winfo event package)
  • Added support for presence payloads (PIDF, RPIDF, CAPS and CIPID schemas)
  • Added XCAP client capabilities
  • Added XCAP server query capabilities (xcap-caps)
  • Added XCAP contacts storage (resource-lists and rls-services)
  • Added XCAP contacts synchronization between multiple instances (xcap-diff)
  • Added XCAP presence policy (OMA org.openmobilealliance.pres-rules)
  • Added XCAP storage for User Icon (OMA org.openmobilealliance.pres-content)
  • Added call history menu (Missed, Received, Placed)
  • Disabled curently unused menu entries and preference options
  • Bug fixes inherited from SIP SIMPLE client SDK
  • Set audio sRTP to be optional by default

Version 0.2.10

September 20th, 2012

  • Added GRUU support (RFC 5627)
  • Fixed handling stream hold edge cases
  • Fixed crash when bogus G722 payload is received
  • Fixed crash on SDP version overflow
  • Fixed engine failure on bogus incoming REFER requests
  • Fixed crash on RTCP decryption when using SRTP
  • Updated Python to version 2.7.3
  • Updated PyQt to version 4.9.4
  • Enable TLS and SRTP labels when appropriate

Version 0.2.7

May 25th, 2011

  • Improved interoperability with OnSIP service
  • Fixed duplicate account detection
  • No longer decode display_name as it's unicode now
  • Adapted to the latest API changes in middleware
  • Fixed string representation of SIP URIs with special characters (SIP Simple)
  • Fixed SDP negotiation on bogus answers (SIP Simple)
  • Reduced SDP size when streams are disabled (SIP Simple)

Version 0.2.6

March 22nd, 2011

  • Fixed exception when NAT type detection is attempted without connectivity (SIP Client)
  • Fixed exceptions when contact URI can't be built for the desired route (SIP Client)
  • Fixed crashes and increased resilience when connectivity is lost (SIP Client)
  • Relax check on SDP origin to increase interoperability (SIP Client)

Version 0.2.5

February 16th, 2011

  • Fixed saving TLS options (SIP Simple)

Version 0.2.4

February 15th, 2011

  • Added support for unicode device names
  • Added menu entry and dialog for joining a conference
  • Restructured main menu
  • Improved DNS resolver capabilities (SIP Simple)
  • Only handle records in the local. domain for bonjour (SIP Simple)
  • Send 500 response if we fail to create incoming invitation (SIP Simple)
  • Fixed race conditions in subscription handlers (SIP Simple)
  • Fixed exception when the session is ended on error conditions (SIP Simple)

Version 0.2.3

December 14th, 2010

  • Detect change of IP address
  • Added web server tools activity indicator
  • Fixed compatibility with older python-qt
  • Made changes to Preferences thread safe (SIP Simple)
  • Fixed TLS transport initialization (SIP Simple)
  • Added DNS resolver autodetection capabilities (SIP Simple)
  • Fixed matching of media codecs on incoming calls (SIP Simple)

Version 0.2.2

November 29th, 2010

  • Fixed detection of audio codecs without a rtpmap line in SDP
  • Fixed exception for MWI NOTIFY without a Message-Account body

Version 0.2.1

November 26th, 2010

  • Allow name and group attributes to be missing when updating a contact
  • Handle bonjour neighbour record updates
  • Updated debian dependency on python-sipsimple
  • Honor the account.sip.always_use_my_proxy setting
  • Fixed opening the create account dialog on first run

Version 0.2.0

November 11h, 2010

  • First Blink QT official release for MS Windows
  • Added the preferences panel
  • Enable inband DTMF by default
  • Disable ICE by default
  • Simplified MWI code and improved its user interface
  • Improve handling of Google contacts
  • Open the dialog for adding the initial account after the main window
  • Switch to new plugged-in device automatically if we have active calls
  • Added transparency for contact icons
  • Added conference contact on first start
  • Many bug fixes in the middleware
  • Adapted to the latest changes in SIP SIMPLE client SDK

Version 0.1.4

September 6th, 2010

  • Save preferred media when creating a contact
  • Fixed broken dependency to python-aplication for non-Debian systems
  • Display 'no new messages' text before getting MWI NOTIFY

Version 0.1.3

September 3rd, 2010

  • Added support for inband DTMF dialing
  • Improved logic for matching contacts to incoming sessions
  • Added pstn prefix setting
  • Fixed enabling Bonjour account item in the menu
  • Added initial MWI support

Version 0.1.2

August 19th, 2010

  • First beta release for Microsoft Windows
  • Switch automatically to the plugged audio device
  • Release notes available at

Version 0.1.1

August 13th, 2010

  • First public release for Debian and Ubuntu Linux
  • Release notes available at
  • Multiple SIP accounts
  • Easy to setup accounts, only the SIP address and password are required
  • Bonjour discovery mechanism
  • Automatic detection of IP address changes
  • TLS Security for both signaling and media
  • NAT traversal using ICE
  • Built-in DNS resolver to by-pass broken implementations in NAT routers
  • Re-INVITE support for adding and removing media streams
  • One-click SIP account sign-up at
  • Integration with AG Projects Multimedia Service Platform
  • Integration with third-party SIP service providers
  • Wideband Audio (G722 & speex)
  • Multiple parallel calls
  • Play hold tone and disconnect tone
  • In-band DTMF support for legacy devices
  • Per account ringtones
  • Silent mode (do not ring on incoming call)
  • Mute microphone
  • Displays packet loss and round trip time
  • Displays selected audio codec and sampling rate
  • Control for input, output and alert audio devices
  • Automatic DTMF mapping between letters and digits
  • Support for entering PSTN numbers and SIP addresses
  • Strip unwanted characters from telephone numbers
  • Redial last call
  • Multi-party conferencing with unlimited number of participants
  • Multiple simultaneous conferences
  • Drag and Drop contacts to conferences
  • Mute individual participants
  • Audio recording
  • Display the caller icon and name retrieved from Address Book
  • Reject calls with 486 Busy or 603 Decline
  • SIP, DNS, MSRP protocol trace to file