NEW - Configurable jitter buffer for each call
NEW - Add record field to add relay call event args - to support configurable screener recording in caller one
NEW - Add support for plain old webrtc call without anywhere, added relevant signalling support
NEW - Change add relay call api to support successive calls on the same device
NEW - Fix FileIn to use a clock based on the sample rate and bytes per sample
NEW - Add support for FileIn, FileOut in key string
NEW - Additional objects to support System.Net serialization issues with .Net core
NEW - Upgrade to latest version of MediaSuite (5)
NEW - Convert to .Net 4.7.1
FIX - Offset byte stream can cause raucous white noise
FIX - RTP decoding can stop under certain circumstances
FIX - Crash bug when transferring starting/stopping calls on devices rapidly
FIX - High CPU usage with multiple Opus calls
FIX - Jitter buffer count excessive warning should be based on frame expiry setting
FIX - Asio input dispose not implemented correctly
FIX - Access violation error during audio frame push when web RTC signalling server updated with active calls
FIX - Prevent anywhere audio pushing whenever ICE connection state is not connected
FIX - Audio Server crash after answering Anywhere call to device
FIX - Anywhere call not working with recent audio info change
FIX - Anywhere call audio broken and delayed
FIX - Merge recent device / rtp / recording fixes into Anywhere calls codeFIX - Web socket session errors on call removal from device
FIX - Anywhere Audio is sometimes distorted
FIX - WebRtc disposal blocking due to marshalling error - cause of memory leak
