Allows the remote party to override payload type for the media we send to it. Payload types are no more overridden locally, and remote settings are used only for the media we sent to the that participant.

Prevents multiple dialing sounds from being played for multiple peers in a single call/telephony conference or for a peer who is being added to a telephony conference when there are other peers there already.

Renames a couple of methods upon request by Emil Ivov. Fixes an endless loop on hangup. Commits work in progress on sound notifications.

Commits work in progress on refactorying the call-related user interface for the purposes of audio and video telephony conferencing over single or multiple protocols and Jitsi VideoBridge.

Corrects media stream stats documentation and a null pointer exception which occurs when the media stream is closed.

Improves sound device selection: automatically switches to new plugged device, remember user preferences in a list.

Commits work in progress on refactorying the call-related user interface for the purposes of audio and video telephony conferencing over single or multiple protocols and Jitsi VideoBridge.

Adds the number of packets discarded by the FMJ packet queue to the "call info" window and to the logs at the end of a call.

Implements a mechanism that allows JMF Controls registered in the  codec implementations to be accessed in MediaStream. Adds a FECDecoderControl interface and uses it to collect FEC statistics in MediaStreamStatsImpl. Adds a "Packets decoded with FEC" line in the call info window.

Change versioning to allow for an easier integration of SVN revision and build number. Removes version revision leaving the version to major.minor[.build-label] where build label may include things like svn revisions, dates, etc.

Adds more options to sound notifications and their config. Sound notifications now can use playback, notification or pc speaker device.

Adds FEC support for the SILK codec. Minor clean-ups in EncodingConfiguration and MediaConfigurationImpl.

Initialize EncodingConfiguration before postInitializeOnce, fixes exception MalformedURLException: Unknown protocol: portaudio when creating ZrtpFortunaEntropyGatherer.

Makes the "toggle video" button in the call window active only if we're ready to send video, and shows an appropriate tooltip otherwise. Same for the desktop sharing button.

Makes MediaServiceImpl#getDefaultDevice return non-null even if no formats are enabled for the device. Updates some of it's usages accordingly. This fixes a problem with SIP audio calls failing when all video formats are disabled.

Changes the EncodingConfiguration instance stored in MediaServiceImpl to one which contains global settings. Changes the names of some methods in EncodingConfiguration and MediaService to make their meaning more clear. Adds storeProperties() methods to EncodingConfiguration, makes EncodingsPanel use them.

Addresses incorrect and incomplete descriptions and documentation regarding libjitsi's examples. Reported by Alexander Fedulov.

Adds support for the IETF Opus codec (patch by Boris Grozev). Fixes a problem that prevented Jitsi from using formats that are disabled by default (pach by Boris Grozev)

Applies a patch from Boris Grozev that allows setting different codec configurations to different accounts. It is now possible to set entirely different codec preference for every account.

Corrects writing DTMF payload (RFC4733) for RTP packets, when the RTP packet length is greater then the fixed header (i.e. with CSRC identifiers).

Fixes issues with video conferencing such as the display of a non-focus participant's video to other non-focus participants and the stopping of the streaming between non-focus participants upon stopping of the streaming of the focus' local video.