Jitsi (previously SIP Communicator) is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo!, Bonjour, and many other useful features. Jitsi supports the ZRTP protocol stack by Phil Zimmerman for encrypted private communications.

Jitsi is Open Source / Free Software, and is available under the terms of the LGPL and performs secure video calls, conferencing, chat, desktop sharing, file transfer, support for your favorite OS, and IM network. All this, and more, in Jitsi - the most complete and advanced Open Source communicator.

Note: Now requires a 64-bit Intel processor.

Security fixes

• Updates otr4j to version 0.22 which fixes a security vulnerability in the AKE
• Stop XML External Entity (XXE) Processing
• Stop XSS with nicknames in the chat window
• Disable the support for SSLv3 in XMPP

New features

• Shiny new support for the IRC protocol
• Proper Gatekeeper Support on OS X 10.10 Yosemite
• Support for Java 8
• Webstart launcher (beta and in source only)
• "In a meeting" status (on Windows with Outlook only)
• DNS packages are now logged as well
• Splash screen on startup
• Adaptive Multi-Rate Wideband (AMR-WB) audio codec, also known as G.722.2

OS X-Specific

• Use an embedded Java 8 runtime
• Use CoreAudio as the default audio system
• Enable echo cancelling by default

Removed support

• MSN protocol
• Google's legacy Jingle
• 32-bit OS X operating systems

Bugfixes/Improvements

• SIP/XMPP
• Try to register more aggressively when dialling a URI on startup
• Plays the DTMF sounds immediately upon pressing the respective buttons; otherwise, playing them sequentially results in sounds which are played/heard many seconds or even a minute after the respective button is pressed.
• Fixed loss of audio of paused/on hold calls with SRTP encryption due to a failure to authenticate RTP packets
• Automatic toggling of the 'On the phone' status (disabled by default)
• Fix ZRTP security status notification when video gets disabled
• Supports signalling Opus in a more interoperable way (with both 1 and 2 channels)
• Brings back recording in WAV format

• SIP
• Handle screen name updates
• Don't write call history record from queue calls (RFC3226 reason cause=200;text="Call completed elsewhere")
• Add support for callto: and tel: URIs
• Fixes an issue with non-responses to INVITEs arriving within a TERMINATED dialog by responding with a 481 Call/Transaction Does Not Exist
• Adds support for NOTIFICATION messages that are not synchronised with the SUBSCRIBE request
• Respect port in registrar-less SIP calls
• Better reconnection handling on TLS connections
• Better negotiation of SDES encrypted calls
• Adds CRLF keep alive method
• Stop keep alive when the server reports a wrong password
• Enables sending of RFC4575 partial notifications by default

• XMPP
• Adds support for undefined Jingle reason codes.
• Correctly parse XMPP addresses with an @ in their resource name
• Support for anonymous logins
• Connection information frame that shows the active server and the encryption ciphers
• Reuses chat thread IDs when sending messages
• Implements XEP-0280 (message carbons)
• Send typing info only after the first message has been sent and only to the resource the messages is actually sent to
• Adds the ability for the OTR plugin to work with resources

• Chat
• Larger history buffer in chat panels
• Does not select already opened multi chat tabs on new messages, just changes the color and adds the number of unread messages
• Prints friendly message when user has its voice revoked and tries to send a message
• Fixes banning chat room members
• Recent messages as contacts
• Updates server side history loading for chat rooms
• Option to automatically open a chat room from is now available from the context menu
• Fixes HTML being shown in OTR messages

• General
• Show the MetaContact's name in search results
• Show the entire chain of X.509 certificates on validation failures
• Does not try to modify metacontact list storage when adding contact to a non persistent group
• Fixes number of packets and lost packets statistics for media stream
• Fixes 'Scratch noise using opus/48000 codec'.
• Displays global status message when available.
• Fixes muting local input for all peers in conference calls created after merging a call into one to one call (used to mute only the peer from the initial call)
• Adds phone numbers to LDAP query searchable attributes
• Changes the ProtocolContactSourceService search for contacts to be case insensitive.
• Displays date or hour in front of recent messages content
• Encode display names to support commas in the call history
• Fixes auto selection of the first contact for search filter
• Fixes issues with renaming groups and moving a user from group to the root group
• Options to customize LDAP queries
• Uses client-to-mixer audio levels as defined by RFC 6464
• ConfigService based on HSQLDB (intended for users with huge configuration files, optional and disabled by default)
• Show UI language selection in current and destination language along with the progress of the translation
• Fixes the support for G.722