Changes since: Audio Server v2 2.9.1.24
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