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