Changes since: Audio Server v2 2.7.0.7
NEW - Add WebRTC components
NEW - Add support for SDP fmtp attribute with OPUS
NEW - Build with latest Sdp library for s=" " fix
NEW - Add TURN support from audio server for WebRTC calls
NEW - Add Relay calls for WebRTC
NEW - Add anywhere endpoint configuration from ini file
NEW - Implement support for more OPUS bitrates
NEW - Add ability for OPUS bitrate to follow what is being sent
NEW - Refactor to take phonebox reference away and move SDP class library to BBCommon.Sip
NEW - Build with latest Sdp library for s=" " fix
FIX - Problems with Opus media format lines in reading/writing SDP
FIX - Anywhere common files not installing properly resulting in Anywhere failures
FIX - Cpu utilization in minutely log entry needs to be divided by number of cores to be accurate
FIX - Sdp differences cause double speed audio on WebRTC call when Firefox makes remote call
FIX - Delay on relay device calls
FIX - Fix reference to BBCommon in Audio.v2 library project
FIX - Slight audio delay issue persisting for WebRTC calls
FIX - WebRtc.dll is not versioned correctly
FIX - Delay building up over time for webrtc calls
FIX - Anywhere segfault caused by answer being set twice on call park
FIX - Prevent Anywhere web socket disconnection
FIX - improve call recording to network paths
FIX - benign error during purging when default ringtone wav is present
FIX - Problems with Opus media format lines in reading/writing SDP
FIX - Cpu utilization in minutely log entry needs to be divided by number of cores to be accurate
FIX - Opus sdp interpretation of channels segment of 'a' line should not affect stereo/mono behaviour
FIX - Build to incorporate positive custom payload number fix for skype and vx handsets
