Libjingle

0.6.11 - Jan 24, 2012
  - Improved ipv6 support.
  - Initial DTLS support.
  - Initial BUNDLE support.

0.6.10 - Jan 11, 2012
  - Support fullscreen screencasting of secondary displays.
  - Add IPv6 support for libjingle's STUN components.
  - Enable SRTP in PeerConnection v1.
  - Bug fixes.

0.6.9 - Jan 09, 2012
  - Enable SRTP in PeerConnection.
  - Bug fixes.

0.6.8 - Dec 22, 2011
  - Add a lot of unit tests
  - Add a lot of older files to base/ and xmpp/
  - Add examples/pcp add examples/peerconnection
  - Improve support for IPV6

0.6.7 - Dec 21, 2011
  - Release new PeerConnection implementation to app/webrtc.
  - Bug fixes.

0.6.6 - Dec 14, 2011
  - Fix support for rtcp multiplexing (aka rtcp-mux).
  - Add more support for FreeBSD and OpenBSD.
  - Add more unit tests to session/phone.
  - Add session/phone/mediarecorder.cc.
  - Fixed httpportallocator tests.

0.6.5 - Dec 8, 2011
  - Add IPv6 support in SocketAddress.
  - Change PeerConnectionFactory inteface.
  - Bug fixes.

0.6.4 - Nov 30, 2011
  - Branch app/webrtc to app/webrtcv1.
  - Add more base unit tests.
  - Add xmllite unit tests.
  - Refactoring and bug fixes

0.6.3 - Oct 26, 2011
  - Add media unit tests
  - Improve OpenSSL support
  - Add SSL unit tests
  - Add DTLS support to SslStreamAdapter
  - Add initial support for media processors
  - Updated WebRTC voice and video engines

0.6.2 - Oct 7, 2011
  - Increase the video rtp buffer.
  - Disable sound system for chromium build.
  - Add basictype.h for NULL.
  - Use the ref counted webrtc ADM/VCM.
  - Add codereview.settings to use the webrtc codereview system.
  - Add MediaSessionDescriptionFactory.

0.6.1 - Sep 15, 2011
  - Add dummydevicemanager.
  - Remove underscores from the files names for app/webrtc folder.
  - Remove PeerConnection OnLocalStreamInitialized callback.
  - Fix webrtcjson.cc numeric locale formatting issue.
  - Don't start playout until the local content has been set.

0.6.0 - Sep 13, 2011
  - Add pub sub support
  - Add unit tests

0.5.9 - Aug 31, 2011
  - Add app/webrtc
  - Add webrtcvoiceengine/webrtcvideoengine
  - Add some unit tests
  - Add XMPP MUC room config classes
  - Update STUN support some more (RFC 5389)
  - Add video output scaling
  - Refactoring and bug fixes

0.5.8 - July 1, 2011
  - Support for loudest speaker detection

0.5.7 - Jun 23, 2011
  - Support for setting MUC display name
  - Update STUN support to RFC5389
  - Handle description-info message
  - New call flag: --debugsrtp

0.5.6 - Jun 2, 2011
  - Improved mac socket server
  - Add IqTask
  - Flush output in examples/call
  - Bug fixes

0.5.5 - May 26, 2011
  - Refactor async sockets
  - Improve MUC joining
  - Add OSX video renderer
  - Bug fixes

0.5.4 - May 13, 2011
  - Support for MUC lookup by name
  - Bug fixes

0.5.3 - May 10, 2011
  - Stream notification and selection.
  - Better XEP-0045 support.
  - Easier to create composite media engines where one part is fake.
  - Make GtkVideoRenderer thread-safe.

0.5.2 - Jan 11, 2010
  - Fixed build on Windows 7 with VS 2010
  - Fixed build on Windows x64
  - Fixed build on Mac OSX
  - Added option to examples/call to enable encryption
  - Improved logging
  - Bug fixes

0.5.1 - Nov 2, 2010
  - Added support for call encryption.
  - Added addtional XEP-166 and XEP-167 features:
    - Call redirection
    - Candidates in session-accept or session-initiate
  - Added support for bandwidth control.
  - Added features in examples/call:
    - bandwidth control on initiate or accept
    - turn on/off SSL
    - control signaling protocol
    - send chat message

0.5.0 - Sep 16, 2010
  - Implemented Jingle protocols XEP-166 and XEP-167.
  - Backward compatible with Google Talk Call Signaling protocol implemented
    in previous versions.
  - Builds on Windows, Linux, and Mac OS X with swtoolkit.
  - Removed GipsLiteMediaEngine.
  - Added video support.
  - Added FileMediaEngine to support both voice and video via RTP dump.
  - Many bug fixes.

0.4.0 - Feb 01, 2007
  - Updated protocol.
  - Added relay server support.
  - Added proxy detection support.
  - Many other assorted changes.

0.3.0 - Mar 16 2006
  - New GipsLiteMediaEngine included to make calls using the GIPS
    VoiceEngine Lite media componentry on Windows.

0.2.0 - Jan 27 2006
  - Windows build fixes with Visual Studio Express project files.
  - Pseudo-TCP support provides TCP-like reliability over a P2PSocket
  - TunnelSessionClient establishes sessions for reliably sending data
    using Pseudo-TCP.
  - A new pcp example application transfers files from one user to
    another using TunnelSessionClient.
  - TLS login support for both example applications.

0.1.0 - Dec 15 2005
  - Initial release.