Version 3.4.0
November 6th, 2015
- Fixed Screen Sharing negotiation on OSX >10.9
Version 3.3.1
November 2nd, 2015
Version 3.3.0
October 19th, 2015
- Fixed broken SIP registrations when network conditions changes
- Fixed compatibility with OSX 10.11
- Fixed detecting aspect ratio of remote video stream
- Fix crash when TLS transport is reset while data is being sent
- Fix opening camera with the right resolution on AVF backend
- Fix enumerate supported resolutions in AVF video backend
Version 3.2.0
June 30th, 2015
- Fixed selecting remote video aspect ratio
- Fixed opening semi-HD cameras of latest 2 inch Macs
Version 3.1.1
June 29th, 2015
Version 3.1.0
June 6th, 2015
- Added VP8 codec
- Added pause/resume for file transfers
- Added call history for bonjour sessions
- Added display name to session history table
- Added encryption field to history session table
- Fixed rendering per contact history items
- Fixed redial Bonjour contact
- Don't play encryption sounds if ZRTP not supported
- Handle double click of bonjour contact in history contacts list
- Improve session history management
- Save session encryption information to history database
- Update history items after call was logged to history
- Show entry in missed calls group if call back failed
Version 3.0.7
March 19th, 2015
- Fixed blocking when switching video calls
Version 3.0.6
March 18th, 2015
- Fixed blocking on video call end
Version 3.0.5
March 15th, 2015
- Fixed File Transfer using drag and drop over video window
- Set default encryption type as opportunistic
- Don't log to history ZRTP sas received from SylkServer
- Fixed setting user agent after upgrade or first start
- Fixed draining the message queue in ChatStream
- Fixed blocking when video stops
- pjsip: fix initial packet loss when using ZRTP
- Run blocking ZRTP operations in the file-io thread
- Allow unicode to be passed as the ZRTP peer name
- Add setting for opportunistic SRTP encryption
- Fix sending initial keyframes when ICE is used
Version 3.0.4
February 26th, 2015
- Use chatroom capability detection to send ZRTP SAS over chat
- Send ZRTP SAS over MSRP chat if proper conditions are met
- Don't save remote ZRTP peername if is focus
- Fixed SMS exception
- Don't print packet loss unless is relevant
- Add 'opportunistic' SRTP key negotiation setting
- Skip system address book when searching for existing presence contact
Version 3.0.3
February 17th, 2015
Version 3.0.2
February 13th, 2015
- Fixed TLS library
- Added audio RX/TX packet loss graph in session info panel
- Fixed Chat encryption bug
Version 3.0.0
February 13th, 2015
- New build 64 bit
- Update SDK libraries
Version 2.8.2
February 10th, 2015
- Fixed Video session bugs
- Fixed chat history replication
- Fixed loading SIP server web page when using multiple accounts
- Fixed auto accepting video for Bonjour calls
- Fixed Opus codec negotiation
- Fix MSRP file transfer bug introduced in previous build
Version 2.8.1
February 5th, 2015
- Refactored ZRTP GUI
- Fixed open/close chat drawer
- Fixed aor formatting of sip address
Version 2.8.0
January 30th, 2015
- Render images sent inline chat stream
- Added control to prevent auto scrolling of logs
- Added chat option to silence GUI notifications if session is not focused
- Improve GUI feedback when video session ends
- Enable Spotify pause
- Capture OTR exception in chat web view
- Fixed adding/removing video stream bugs
- ZRTP fixes
- Video fixes
Version 2.7.3
November 1st, 2014
- Fixed app signing with Apple developer ID
Version 2.7.2
October 29th, 2014
- Fixed ZRTP video sessions and enable ZRTP by default
Version 2.7.1
October 27th, 2014
Version 2.7.0
October 25th, 2014
- Added ZRTP encryption
- Fixed SMP identity verification for encrypted Chat sessions
Version 2.6.0
October 18th, 2014
- Change naming convention for screen capture filenames
- Added sending screenshot to video window
- Allow scaling down of video window
- Improved contextual video menu and video transitions to and from audio drawer
- Don't transit to full screen when we detach the video window
- Close video devices when program ends
- Fix deadlock when closing video devices
- Use SSLv23 method for TLS
Version 2.5.5
October 14th, 2014
- Added first Dutch translation
- Added video view to audio drawer
- Added more granular settings
Version 2.5.4
October 8th, 2014
- Change moment of selecting search box field as first responder
- Customize GUI close delay
- Fixed printing chat disconnect message
- Refresh devices when waking up from sleep
- Reload only chat sessions that had actual messages
- Print hint for how to unlock stuck camera
- Don't mangle html payloads
- Hide video cancel button after hangup
- Fixed printing account registration status
- Improve chat disconnect message
- Use thin dividers in chat drawer view
- Fixed handling video view in chat window when call fails
- Set stream status to failed if session fails
Version 2.5.3
September 30th, 2014
Version 2.5.2
September 25th, 2014
- Fixed bringing my video window on top
Version 2.5.1
September 25th, 2014
Version 2.5.0
September 18th, 2014
- Improved Video sessions
- Fix handling initial INVITE requests without SDP
- Fixed handling re-invites in some cases
- Fixed screen sharing handling
- Added icons for indicating account TLS transport
- Properly reregister / resubscribe / republish account after transport disconnects
- Fixed duplicate Missed call section in history menu
- Save Echo Cancellation per combination of audio devices
- Fixed copy/paste from new contact request
- Handle initial INVITE with no SDP offers
- Allow auto-accept screensharing in a re-Invite only if the contact is trusted
- Fixed time offset calculation for missed calls
- Add notifications when TCP/TLS connections break
- Open built-in remote screen viewer when clicking a corespondent url in the chat window
- Automatically open chat drawer only when the remote party is focus
- Improve the handling of SIP transports failures
Version 2.4.0
June 16th, 2014
- Added Portuguese translation
- Update Spanish translations
- Update Romanian translations
- Capture exceptions for divisions
- Fixed check for enabling printing
- Add sleeping icon for contacts
- Fixed time offset calculation
- Add notifications when connections break
- Reset registration status when transports fail
- Reset registration status imediately when disabled
- Open built-in remote screen viewer when clicking a corespondent url in the chat window
- Automatically open chat drawer only when the remote party is focus
- Added explicit support for Blink Chat stream features
- Added support room sample contact
- Improve the handling of SIP transports failures
Version 2.3.0
June 8th, 2014
- Update Spanish Translations
- Load last week by default in History viewer
- Group Chat and SMS together in History viewer
- Added Organization attribute to contacts
- Enable SIP and DNS traces by default
- Put default video values on first position of popup menu items
- Update screen sharing with latest API
- Refactor code to detect supported media per contact
- Don't send presence UI notifications during transitory failures
- Improved logging of failed SIP transports
- Added Presence sound notifications
- Make Chat toolbar customizable
- Play sound when taking video snapshot
- Improve resilience against network failures
- Fixed parsing html chat payloads
- Reregister account after sip transport failure
- Added GUI elements for Voicemail server notifications
- Remove growl notifications
- Handle \ espace character in chat sessions
Version 2.2.0
May 30th, 2014
- Added screenshot action to video control panel
- Added setting for enabling media detection from presence
- Save user agent name after start
- Turn off SMS by default
- Add contextual menu to remove video window
- Display hold status in title bar of video session
- Fixed enabling start session actions on search results
- Added language preferences
- Update translations
- Display localized version of On The Phone presence note
Version 2.1.0
May 22nd, 2014
- Added support for multiple video cameras
- Added Spanish Translation
- Improvements in ICE negotiation
- Improvements in Chat sessions
- Several big fixes in the middleware
Version 2.0.0
May 6th, 2014
- Added Video sessions
- Enhance chat progress indicator
- Improve contextual audio menu layout
- Implement adding multiple streams at the same time
- Use ICE and SRTP by default
- Don't switch automatically to input only device
- Handle reinvite failure if session is not in legal state
- Fixed reusing conference screen sharing
- Fixed opening file transfer window after transfer finished
- Don't fallback to alternative route for 600 codes
- Don't print OTR negotiation timeout as it may succeed later
- Fixed stopping OTR negotiation progress indicator
Version 1.9.1
March 19th, 2014
- Save early if remote party is focus
- Don't reset mediastream_started
- Improve OTR negotiation
- Mark previous sent messages without response as failed
- Fixed widget sizes
- Stop the timer when cancelling photo picking
- Fixed time replication from server history entries by converting to utc
- Decode URLs received from external apps
- Adapt to API changes in AudioStream recording
Version 1.9.0
March 9th, 2014
- Added network bandwidth utilization for audio calls
- Added menu item to open last Instant Messages window
- Added option to strip digits from PSTN numbers
- Fixed default of answering machine setting
- Fixed detecting session state
- Improve debug logging of sessions
- Detect pending hold requests to disable session buttons accordingly
- Improved detection of session status for enabling session action buttons
- Retry alternative route in case of server failure
- Improve visual and audible feedback when taking snapshot with camera
- Save session start and end time in UTC
- Show progress indicator when adding audio from chat window
- Refactor changing presence state
- Start a new session from scratch if proposal fails
- Fixed selecting presence from history
- Handle non ascii characters
- Improve chat system messages
- Improve engine connection logs
- Fixed marking chat messages as failed if stream did not start
- Fixed alert_url handling
- Improved logging of account registration status
- Imediately fail to start session if we have no IP address
- Imediately fail sending SMS if we have no IP address
- Improved logging of network changes
- Fixed account registration status string when no IP address is available
- OTR fixes for SMS messages
- Strip spaces from URIs in contacts
Version 1.8.0
February 8th, 2014
- Added OTR encryption for SMS sessions
- Improved handover between different IP networks
- Print when network conditions changed
- Bring main window on top when clicking on the system menu icon
- Discard availability UI notifications for myself
- Improve anonymous user detection
- Added option to conference menu to tell when participants list has changed
- Don't bring up file transfer window when sending screenshots
- Fixed post dial dtmf check
- Update chat window title
- Improve logging of SMS sessions
- Improved detection of OTR negotiation failure
- Always send own icon after starting a chat session
- Renamed Latency to Network Latency
- Divide RTT value by 2 for audio latency display
- Enable more advanced options for sip2sip accounts
- Don't repeat audio quality log messages
- Get chat window out of the way when sending screen shots
- Improve external alert description
- Allow alert URL to be any type not just http
- Don't hide advanced preferences sections for sip2sip target
Version 1.7.0
January 29th, 2014
- Fixed deconding utf8 in system chat notifications
- Float alert panel on top of screen saver
- Highlight DND accounts with red
- Fixed starting ringtone related tasks
- Log when starting session from call transfer request
- Check if routes exist
- Don't render codec in audio status if call has been transfered
- Removed redundant actions
- Refactored how the streams are resused
- URLify strings found html paragraphs
- Fixed handling failed outgoing pulll file transfer
- Added support for sending rich text using html
- Fix indent of log line
- Shorten chat failure message
- Strip sip: from sip next hop in audio status
- Added font size control for Chat windows
- Don't play local ringtone if early media is active
- Show encryption status for history messages in SMS window
Version 1.6.0
January 17th, 2014
- Show Activity tab on logs window load
- Fixed stopping outgoing ringtone
- Print connectivity failure in sip registration status for some PJ error
- Don't require OTR for Bonjour sessions
- Improve parsing of external URLs
- Allow dragging folders over audio session
- Fixed rendering incoming image file transfer in chat window
- Allow dropping folders over chat window for file transfer
- Log IP address change
- Fixed displaying shared conference files when switching chat tabs
- Stick chat messages from same recipient together
- Add system address book at the end of the group lists when searching matching contact
- Don't print connection system messages for replayed history
- Allow selecting directories in open file selection dialog
- Don't print OTR encryption errors in chat window
- Refactor purging temporary files
- Print registration details once
- Allow transfer of entire folders
- Change some log lines to debug level
Version 1.5.1
January 6th, 2014
- Fixed init AB contact if no uris available
- Fixed updating encryption icon for pending messages
- Send UI presence notifications only when status become available
- Print session call id when starting
- Don't require presence to enable start chat menu item
- Fixed contextual menu item for anonymous history entries
- Show voicemail entries for anonymous callers in history groups
- Added toggle Outgoing calls group to contextual menu
- Fixed cancelling adding audio from chat window
- Stop ringing if re-invite failed due to illegal session status
- Fixed sending pending outgoing chat messages after session has started
Version 1.5.0
December 29th, 2013
- Fixed init AB contact if no uris available
- Purge temp snapshot files on exit
- Avoid flipping audio status after stream ended
- Remove contact from online group when subscribe for presence is disabled
- Added send snapshot from camera to chat sessions
- Fixed OTR negotiating bugs
- Properly capture exceptions when creating a new SIP account
- Fixed setting encryption lock for sent messages without encryption
- Added option to automatically send anonymous calls to answering machine
- Use 101 as the telephone-event payload type
- Fix exception in Presence subscriptions
- Updated opus codec to version 1.1
- Don't set last active timestamp automatically
Version 1.4.0
November 21st, 2013
- Added DND until call is ended
- Added copy/paste from Watcher list to Contact list
- Added Jabber and all Email addresses to AB contacts
- Added option to skip contacts sync with server when adding existing account
- Improve Chat handling of HTML payloads
- Added manual sorting for ContactURIs belonging to a Contact
- Enable audio button only if session is not in progress
- Disable presence if contact sync is disabled during enrollment
- Adjusted to latest middleware changes related to stream addition/removal
- Fixed sorting SIP addresses when droping file over contact
- Lookup icons in blink presence contacts only
- Handle multiple URIS when create presence contact from another contact
- Print job title and organization for AB contacts
- Use first SIP URI as name if missing for AB contacts
- Search AB note and job title
- Automatically process , in phone numbers as DTMF delimiter
- Fixed migrating version for session history table
- Fixed outlet alignment
- Print next hop in audio tile
- Allow deletion of contacts in OnlineGroup
- Use ICE by default
- Fixed reloading history enties after database query has executed
- Fixed restoring contacts from backup
- Don't notify subscriptions to myself
- Mark SMS messages as failed if DNS lookup fails
- Always show logs button
- Fixed debug window inconsistencies
- Fixed log line
- Fixed debug window outlets
Version 1.3.3
October 13th, 2013
- Fixed debug window inconsistencies
- Fixed path for call transfer xib
- Fixed exception during enrollment
Version 1.3.2
October 7th, 2013
- Added chat privacy label
- Added per session/contact chat history storage option
- Added a safety timer for music application muting
- Save use_speech_recognition in blink settings
- Improve settings descriptions
- Refactored logging to work with latest middleware changes
- Allow starting audio call from Chat window
- Honor remote preference for not logging to history database
- Fixed DNS lookup blockiness
- Fixed crash when in-dialog request fails to be sent within a subscription
- Fixed memory leak by initializing the handler after the stream initialized
- Properly handle mutex creation failures
Version 1.3.1
September 27th, 2013
- Refactored toggling visibility of collaborative editor
- Update chat window tooltips
- End session when re-Invite fails for last stream
- Improve Chat SMP OTR window behaviour
Version 1.3.0
September 13th, 2013
- Added SMP fingerprint verification for OTR chat sessions
- Several OPUS codec related fixes
- Use single global c line when creating SDP
- Don'r render system message when cancelling Chat sessions
- Show loading indicator in Chat view when connecting session
- Don't render OTR raw Chat messages
- Renamed audio encryption menu items for clarity
- Don't duplicate system chat message when stream fails
- Fixed starting when no account is selected
- Fixed marking failed Chat messages
- Display logging to database status for chat messages
- Fixed updating status of OTR encryption lock
Version 1.2.1
September 2nd, 2013
- Mark pending outgoing chat messages as failed if the DNS lookup fails
- Fixed reusing audio tile after one sip route fails
- Render encryption lock for each chat message
- Use white font for failed messages
- Fixed processing related messages
- Fixed display of packet loss in audio tile
- Don't atempt using OTR with a remote focus
- Used a closed red lock when chat is encrypted but fingerprint has changed
Version 1.2.0
August 27th, 2013
- Added Chat OTR support
- Fixed toggling visibility of all special groups
- Start ringer cleanup timer after application has started
- Improved wording of SRTP menu items
- Update SRTP in info panel
- Refactored audio encryption user interface
- Fixed detecting when music must be paused
- Make Chat drawer audio status consistent with audio tile status
- Fixed position of call transfer contextual menu
- Fixed detecting when music has been paused
- Added pause music setting
- Change chat connection messages
- Print more concise errors in chat window
- Simply LDAP queries
- Handle changes in System Address Book incrementally
- Don't set default DTMF delimiter
- Load AddressBook entries in a dedicated thread
- Fixed resetting chat controller when re-using it
- Print nicely TLS related errors
- Fixed updating audio tile info
Version 1.1.1
August 8th, 2013
- Reload model after deleting policy entry
- Fixed calling logging function
- Skip connecting to chat replication server if chat replication password not set
- Don't add chat to replication journal if chat replication password not set
- Handle hanging ringing sessions with a clean up timer
- Auto select authorized account for session to XMPP contacts
- Increase GUI responsivness when session ends
- Don't try publish after application ended
- Fetch last 10 entries in history menu
- Don't refresh history menu after every call
- Hide end chat delimiter when replaying Chat history
- Don't print chat session delimiters when replaying SMS history
- Disable history scolling for Bonjour accounts
- Update history menu section titles
Version 1.1.0
August 5th, 2013
- Added elastic scrollback in time for Chat sessions
- Added elastic scrollback in time for SMS sessions
- Added search text capability within Chat and SMS sessions
- Added setting to enable the Answering Machine in alert panel
- Load history messages from all uris belonging to a contact
- Don't log duplicate database entries error
- Change background colors of failed chat messages
- Log chat errors through its associated session controller
- Fixed sending pending messages after connecting chat stream
- Send own icon only for Bonjour chat
- Improved navigation of history entries by date
- Fixed removing outgoing ringtone when using failure route
- Fixed removing on_hold ringtones for failed sessions
- Fixed initialisation of collaboration editor and display
- Fixed selecting missed calls entries
- Render voice messages recorded by answering machine in contact contextual menu
- Renamed Voicemail to Answering Machine
- Reload only the last ten file transfers by default
- Reworked mute music applications using official Scripting Bridge API
- Use night volume settings for speech synthesizer
- Don't auto open chat window at startup for replicated messages
- Fixed crash on SMS window dealloc
Version 1.0.2
July 31st, 2013
- Don't render replicated chat messages older than 2 hours
- Reload last 10 file transfers by default
- Added scrollback in time for chat sessions
- Added scrollback in time for sms sessions
- Rename Voicemail to Answering Machine
- Render voice messages in history contact
- Render voice messages recorded by answering machine in contact contextual menu
Version 1.0.1
July 24th, 2013
- Fixed reset of Acoustic Echo Canceller
- Added add/remove Chat to audio session tile
- Close chat stream when closing tab without ending the session
- Finish session when click on disconnect chat button
- Remove jitter buffer from session info panel
- Replace preferred media with pop up button
- Fixed printing duplicate sip code
Version 1.0.0
July 22nd, 2013
Client Features
- SIP2SIP account and Bonjour Discovery
- Acoustic Echo Cancellation
- Ultra-wideband audio (OPUS codec)
- Wideband audio (G.722 and Speex codecs)
- PSTN audio (G711, GSM and iLBC codecs)
- Play/pause iTunes, VLC and Spotify
- Multiparty Conferencing for all supported media
- Keychain password storage
- Voiceover accessibility support
- Address Book contacts integration
- Answering Machine and Auto-Answer
- Dialpad, DTMF, Hold, Transfer and Recording
- Many Call Automation Features
- Presence (RLS and XCAP)
- Instant Messaging (MSRP protocol)
- File Transfers (MSRP protocol)
- Contacts Management (XCAP protocol)
- Directory Services (LDAP protocol)
- Screen Sharing (VNC over MSRP)
- Interoperability with XMPP domains
- History Viewer for all media types
Service Features
- Multiparty Conferencing for all supported media
- Contacts sync between multiple devices
- Presence sync across multiple devices
- SMS sync across multiple devices
- MSRP chat sessions sync across multiple devices
- Access to SIP server account settings web page
- PSTN termination (paid option)