Changes since: 2.11.1.16
NEW - Add helper to detect default communications devices
NEW - Pass Anywhere WebRTC ICE status and call stats to
Talkshow
NEW - Support for ICE restarts for Anywhere reconnections
NEW - Music on hold audio provided by a SIP device
NEW - Enhance Anywhere call quality log with call ending
reason + session guid
NEW - Improve WAV file playback and implement for device
inputs with default path to audio folder
NEW - Allow devices with no physical sound output
NEW - Use RFC2833 for sending DTMF for Audio Server and
Softphones instead of generated tone
NEW- Update references to latest components and add Opus Ogg
file parser and conversion
NEW - Voicemail style device recording
FIX - Crash issue relating to Anywhere call stats
FIX - Ensure recordings that stop when they reach the
maximum length are made available to clients
FIX - Anywhere connection object not releasing resources
causing memory leak
FIX - If no path is specified for file MOH then use default
audio folder
FIX - Jitter buffer losing packets on Seq rollover
FIX - On marker bit received our packet interval should be
recalculated and data transfer / buffer size operations adjusted accordingly
should it change
FIX - RTP Sequence number not correctly rolling over and
always starting from 0
FIX - Improvements to WDM device reliability to prevent of
crashing at start of call and allow audio to continue if device is disconnected
and reconnected.
FIX - Call recording not working in Automatic mode
FIX - No caller audio after very short ringing period
FIX - Improve log file rotation and purging
FIX - Prevent crash caused by RTP decoder
FIX - Multicast group rejoin/join issues with rapid fire
park cycles
FIX - Corrupt ringtone audio played to RTP calls
FIX - Logged error if File Transfer port hit by scanner
FIX - Problem with caller audio if packet marker with no
payload received
FIX - Remove timeout causing delays when calls are removed
from device/hold
FIX - Unable to reconnect to server after keepalive timeout
