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.

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 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.

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.

Fixes issues with video conferencing such as a failure of the local video to be displayed and a limitation of the conference focus with respect to requiring it to capture video as well.

- Works on removing the video of a video conference participant from display on the remaining non-focus participants. - Tries to fix a crash in PortAudio (often seen at the end of video conference calls).

Adds SDES for XMPP. Works with Jingle and GTalk (only for gmail web app, doesn not work on "talk" for android).

Fixes an issue which would leak at least one Thread per call which would endlessly send out RTCP packets.

Fixes a failure to keep the aspect ratio of the video when switching to full screen. Attempts to remedy a multi-video display when switching between camera and desktop streaming. Tries to fix a crash in the QTKit video capture on Mac OS X. Works on addressing a unnecessary remaining of the camera on Mac OS X open.

Re-applies the ideas of r9718 with a refined implementation on the PulseAudio Renderer in order to reflect changes to the user's choice with respect to the default playback device while in a call.

Temporarily reverts to the PulseAudio Renderer implementation to a pre-r9718 state to fix freezes reported by Vincent Lucas and Kertesz Laszlo on the dev mailing list while further looking into the cause of the problem.