| Libjingle |
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| 0.6.12 - Feb 07, 2012 |
| - PeerConnection client for windows. |
| - Bug fixes. |
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| 0.6.11 - Jan 24, 2012 |
| - Improved ipv6 support. |
| - Initial DTLS support. |
| - Initial BUNDLE support. |
| - Update Jingle protocol to multistream. |
| - WebRTC Bug fixes. |
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| 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. |
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| 0.6.9 - Jan 09, 2012 |
| - Enable SRTP in PeerConnection. |
| - Bug fixes. |
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| 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 |
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| 0.6.7 - Dec 21, 2011 |
| - Release new PeerConnection implementation to app/webrtc. |
| - Bug fixes. |
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| 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. |
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| 0.6.5 - Dec 8, 2011 |
| - Add IPv6 support in SocketAddress. |
| - Change PeerConnectionFactory inteface. |
| - Bug fixes. |
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| 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. |
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| 0.6.0 - Sep 13, 2011 |
| - Add pub sub support |
| - Add unit tests |
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| 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 |
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| 0.5.8 - July 1, 2011 |
| - Support for loudest speaker detection |
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| 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 |
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| 0.5.6 - Jun 2, 2011 |
| - Improved mac socket server |
| - Add IqTask |
| - Flush output in examples/call |
| - Bug fixes |
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| 0.5.5 - May 26, 2011 |
| - Refactor async sockets |
| - Improve MUC joining |
| - Add OSX video renderer |
| - Bug fixes |
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| 0.5.4 - May 13, 2011 |
| - Support for MUC lookup by name |
| - Bug fixes |
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| 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. |
