Audio Server 2.11.1.91 (BETA/FROG)

Changes since: 2.11.1.68

NEW - Allow ringing to be played during active conferences

FIX - Update third party components

FIX - Update to framework 4.72

FIX - Update Newtonsoft JSON reference

FIX - Update OWIN web hosting references

FIX - Issues decoding first packet of segment and determining updated packet duration

FIX - Memory leak at end of call

FIX - Back out changes for dynamic jitter buffer and recalculate packet duration from marker bit segments, and improve logging.

FIX - Anywhere calls not answering due to BBCommon.Rest issues

Audio Server 2.11.1.68 (STABLE/FROG)

Changes since: 2.11.1.16

 NEW - Audio server update to retry if its a new version is available and the download fails

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

Audio Server 2.11.1.16 (STABLE/FROG)

Changes since: 2.11.1.0

NEW - Support for ICE restarts for Anywhere reconnections

NEW - Enhance call quality log with call ending reason + session guid

NEW - Music on hold audio provided by a SIP device

NEW - Add helper to detect default communications devices

NEW - Pass Anywhere WebRTC ICE status and call stats to Talkshow 

FIX - Trap errors when retrieving default communications device

FIX - Websocket connections stuck trying to send in an aborted state

Audio Server 2.11.1.0 (STABLE/FROG)

Changes since: 2.11.0.23

FIX - Websocket client connection object not releasing pipe threads

FIX - Ensure recordings that stop when they reach the maximum length are made available to clients

Audio Server 2.11.0.23 (BETA/FROG)

Changes since: 2.10.1.49

NEW - Enable relay calls between audio servers

NEW - Audio server update to retry if its a new version is available and the download fails

NEW - Re-enable stats logging in Anywhere

FIX - Crash due to collection modified concurrency issue in webRtc dll relating to stats

FIX - Access violation crashing audio server

FIX - ReportContext object memory leak in WebRtc

FIX - Frog build not including correct Newtonsoft binding redirect entry in config file

FIX - If an Anywhere call is parked & dropped the Audio continues to flow to the browser

Audio Server 2.10.1.49 (STABLE/FOX)

Changes since: 2.10.1.37

NEW - Implement getStats call and log / report webRtc call quality

NEW - Audio server update to retry if its a new version is available and the download fails

NEW - Re-enable stats logging in Anywhere

NEW - Post Anywhere call stats to Anywhere server per call

NEW - WebRtc call stats logging

FIX - Anywhere Rest API calls not using proxy

FIX - Moh stereo split not working for channels 1 + 2 WDM from a stereo input

Audio Server 2.10.1.37 (STABLE/FOX)

Changes since: 2.10.1.23

NEW - Add support for REST API file (recording) transfer

NEW - Add REST API compatibility check to legacy TCP file (recording) transfer.

NEW - Bind rest API to all network interfaces

FIX - Installer missing DLLs to support REST

FIX - No MoH audio

FIX - CTD from log writer being accessed before initialised when timer resolution override is used

FIX - High CPU usage when device recording enabled

FIX - Fixes to ensure stereo and mono channel combinations are processed properly

FIX - Some soundcards cannot accept mono channel streams

FIX - Fix inaccurate capture of recording length

FIX - Truncated recordings due to file streamwriter backlog

FIX - Allow option to override clock/timer resolution in windows

FIX - Split From Stereo WDM inputs not  isolating the correct channel

Audio Server 2.10.1.23 (STABLE/FOX)

Changes since: 2.10.1.9

NEW - Make the maximum call recording limit configurable
NEW - Sign binaries and installer
NEW - Reduce Anywhere signalling for call moves within an audio server
NEW - Add logging to troubleshoot delays and analyse calls better

FIX - Revert buffer overrun code from previous build
FIX - Long delays experienced on some telco calls on anywhere-enabled
FIX - Tighten RTP sending / encoding timings
FIX - Recording duration values passed in recording available message could be incorrect due to UTC time issue
FIX - Call recording completion not clearing affected calls list
FIX - Prevent webrtc call ending when in disconnected state
FIX - Fix to pick up latest websocket session code fixes
FIX - Fix MoveCall for MOH WDM device
FIX - Half-speed audio on Anywhere calls
FIX - Anywhere audio not send in completed state
FIX - Sporadic MoH multicast error
FIX - Disk write delays when closing off recordings can cause latency on subsequent calls
FIX - Improve logging in device object for anywhere / telco
FIX - Memory growth of audio server during Anywhere calls
FIX - Multicast MoH receive error after udp client dispose

Audio Server 2.10.1.9 (STABLE/FOX)

Changes since: 2.10.1.1

NEW - Add time limit to calls to avoid filling disc

FIX - Fix for large offset or reset of Rtp sequence numbers
FIX - Reduce log level of anywhere call entry
FIX - Problems with backup server connection when using host-name instead of IP address

Audio Server 2.10.1.1 (STABLE/FOX)

Changes since: 2.10.0.66

NEW - Complete rewrite of jitter buffer and associated interaction from RtpCall objects and clock changes
NEW - Auto Updater
NEW - Ensure recording cannot be started again if its already started resulting in orphan files in the tempAudio folder
NEW - Implement comfort noise generation for unrecognised inbound payload packets

FIX - MOH not working for anywhere calls
FIX - MoH audio encoding issue with Opus calls
FIX - Anywhere inbound audio bug when jitter occurs
FIX - Moving call from MoH to regular device happening without any delay hitting audio issues
FIX - Fix startup logic to supply silence where packets are not ready as buffer is filling
FIX - Silence not inserted into call recordings when packets are lost
FIX - WaveIn initialiser using output index for input index

Audio Server 2.10.0.66 (BETA/FOX)

Changes since: 2.10.0.58

FIX - Fix anywhere calls to use TURN over tcp

Audio Server 2.10.0.58 (BETA/FOX)

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

Audio Server 2.9.1.24 (STABLE/EAGLE)

Changes since: 2.9.1.20

FIX - High CPU usage with multiple Opus calls
FIX - Offset byte stream can cause raucous white noise
FIX - RTP decoding stops

Audio Server 2.9.1.20 (STABLE / EAGLE)

Changes since: Audio Server 2.9.1.9

FIX - Access violation error during audio frame push when web RTC signalling server updated with active calls
FIX - Prevent audio pushing whenever ice connection state is not connected
FIX - Audio Server crash after answering Anywhere call to device
FIX - Asio input dispose not implemented correctly
FIX - Crash bug when transferring starting/stopping calls on devices rapidly
FIX - Web socket session errors on call removal from device
FIX - Anywhere call audio broken and delayed
FIX - Glitching whilst recording
FIX - Rtp out buffer not flushing during surges after delayed audio from soundcard
FIX - Retrieval from jitter buffer should discard in the case of a delay followed by a flood of source packet sends

Audio Server v2 2.9.1.9 (BETA)

Changes since: Audio Server v2 2.9.1.8

NEW - Log a warning when silence is sent to ASIO out buffer
NEW - Add verbose Asio log entries for buffer swap analytics

FIX - Asio out buffer issues with buffer swap times relative to incoming RTP audio times
FIX - Jitter buffer not working correctly
FIX - Broken audio on anywhere call

Audio Server v2 2.9.1.8 (BETA)

Changes since: Audio Server v2 2.9.0.80

FIX - Core fixes to sound in flow, device stream, Rtp send stream and device recording and reduction of CPU usage with internal byte buffers
FIX - TURN server not initialising correctly for anywhere calls

Audio Server v2 2.9.0.80 (BETA)

Changes since: Audio Server v2 2.9.0.78

NEW - Change WebRTC to version 71
FIX - Fix memory leak with webRtc threads

Audio Server v2 2.9.0.78 (BETA)

Changes since: Audio Server v2 2.9.0.67

FIX - WebRtc disposal blocking due to marshalling error - cause of memory leak
FIX - Add web rtc configurable frame expiry, set sensible default
FIX - Updated anywhere API to latest version
FIX - MoH devices need to release packets at consistent 20ms intervals

Audio Server v2 2.8.1.30 (STABLE)

Changes since: Audio Server v2 2.8.1.27

FIX - Allow differing payload type values for inbound and outbound RTP
FIX - Linphone issues with Opus call quality
FIX - Rebuild to for opus constant bitrate fix in SDP library

Audio Server v2 2.9.0.67 (BETA)

Changes since: Audio Server v2 2.9.0.65

FIX - Allow differing payload type values for inbound and outbound RTP