This is a quality update with bug fixes only.
The upcoming major new versions are planned to be released in June with many new features and improvements.

• new: notification objects (events instead of strings)
• new: API to receive the media streams: API_GetMedia()
• new: blacklist and whitelist
• new: full video support
• new: video related API
• new: calltype function parameter for Call and Accept
• new: external video in/out streaming
• new: handle Replaces
• new: multiple separate simultaneous conference calls
• new: built-in GUI icon and right click popup
• new: query log file path (API_GetLogPath)
• new: notifications about incoming/outgoing SIP messages
• new: sendblocknotifications parameter and BLOCK notifications
• new: re-register on no keepalive answer (reconnectonnokeepalive)
• new: icon for the built-in user interface
• new: configurable retransmission timeout (timer3)
• new: handling maxptime
• new: configurable listen address for the built-in API (wpapilistenip)
• new: API_IsTerminated
• new: sendearlyack setting
• new: rejectcallsfromunknown setting
• new: disccode, userrejectdisccode and blockrejectcode settings
• new: p_referred_identity, p_asserted_identity, remote_party_id and privacy settings
• new: beeptype, systembeeptype and beepfrequency settings
• new: music on hold (MOH)
• new: security and strictness settings and documentation
• changed: rtpkeepaliveival defaults to 25000 (25 seconds)
• improved: ipv6 handling, ipv4/ipv6 auto-detect
• improved: audio device enumeration on linux systems
• improved: re-invite
• improved: SDP parsing (both speed and correctness)
• improved: connection recovery
• improved: endpoint lookup performance optimizations
• improved: MWI parsing
• improved: presence XML parsing
• improved: presence better parsing for incoming notifications
• improved: presence retry after failure
• improved: chatreports
• improved: audio streaming
• improved: attended call transfer
• improved: call hold state machine
• improved: loading environment variables
• improved: failover (to other transport protocol, backup server, SRV DNS, re-init)
• improved: dynamic codec payload number allocations
• improved: new call during SIP stack startup
• improved: RTP source change handling
• improved: TCP API
• improved: sample rate calculations
• improved: thread safety
• improved: audio mixer info
• improved: not using audio threads when local audio device is disabled
• improved: keeping the correct RTP stream after SSRC
• improved: open/close audio device from separate thread (java audio bug workarounds)
• improved: opus codec properties in the SDP
• improved: RTP keep-alive
• improved: DTMF dynamic payload handling
• improved: ringincall and beeponincoming handling
• improved: clear non-persistent local cache on version change
• improved: delayed initialization
• improved: IMS 3GPP
• improved: API_SendSIPMessage
• improved: SIP message decoding
• improved: switch between transport protocols
• fix: accept linux \n only line separator for the config/ini file
• fix: answer for OPTIONS requests in call sessions
• fix: log queue overflow with high log levels
• fix: opus wideband vs narrowband mismatch in some circumstances
• fix: prevent SIP stack re-init at new call init
• fix: loading the correct AEC configuration
• fix: socket thread terminate on failed recv
• fix: don't reject register if no password configured
• fix: don't show PRESENCE as MESSAGE
• fix: handle different domains with multiple accounts
• fix: MD5-sess auth
• fix: ringtone doesn't stop at call connect on certain hardware/drivers
• fix: double add tech prefix
• fix: other minor bug fixes and improvements

This is a quality upgrade with bug fixes only.

• new: RTP extension (Section 5.3.1 of RFC3550)
• new: IPv6 support in all modules (transport/SIP/SDP/media)
• new: unique IM ID
• new: SIPREC implementation (RFC 7866)
• new: real-time local streaming option
• new: ED-137 (implementation of the ED-137 specification as VCS from ground to air radio)
• new: In-Reply-To SIP header handling
• new: intelligent max udp packet length calculation
• new: Authentication-Info
• new: redial (autoredial parameter)
• new: API over websocket (beside UDP, TCP and HTTP)
• new: API_RTPHeaderExtension (Set RTP header extension for the RTP packets)
• new: API_SetLineParameter (set certain parameter for a specific line)
• new: API_GetLineParameter (get line specific parameter)
• new: API_SetSDPField (set custom SDP field globally or for a line)
• new: API_GetSDPField (get custom SDP field)
• new: API_ED137PTT (ED-137 Push to talk)
• new: capability to spawn multipe signaling sockets on demand with different servers
• new: customsdpfield and customsdpmediafield parameters
• new: rtpextraheader and rtpextraheadernotify parameters
• new: keepoldcustomvalues and networkchecks and rtpkeepalivemultiplier parameters
• new: mainlogicwaitmultiplier parameter (might be used when real time operations are required)
• new: natopenpacketsalways parameter (send NAT open packets)
• new: sendmedia_seq parameter (insert sequence number for local stream; useful for debugging)
• new: changesystemsettings parameter (set to 0 to prevent changing JVM properties by System.setProperty)
• new: logquesize parameter (log queue max length multiplier; default is 1)
• new: answerwithallcodec parameter (list all supported codec in the SDP answer)
• new: siprec_src, siprec_src_start, siprec_srs_uri, siprec_srs, siprec_srs_hide SIPREC related parameters
• new: ed137, defptt, processmutedrtp, pttid, maxforwards, disconed137rtptimeout ED-137 related parameters
• new: auto detect available ciphers (don't try to use missing ciphers)
• new: INFO notification
• new: RTPE notification
• new: REGISTER notifications
• new: handle R2S-KeepAlivePeriod and R2S-KeepAliveMultiplier
• improved: endpoint reinit on register failure
• improved: audio device pre-init and caching
• improved: dns record cache
• improved: RTP header parsing
• improved: RTP timing accuracy
• improved: mute and hold
• improved: operation with no audio device (streaming only)
• improved: headless mode
• improved: network availability detection
• improved: merge (extra) SIP headers
• improved: local media streaming
• improved: auto guess best SDP local ip
• improved: 3GPP SMS
• improved: better handling of 513 Message Too Large responses
• improved: remove not negotiated codec's
• improved: release session specific global objects on session destroy
• improved: GetLocalAOR
• improved: handling incorrect packet size for GSM codec multiple frame per packet
• improved: use the correct local listener port also for logic SIP URI's
• improved: SIP stack re-init
• improved: socket (UDP/TCP/TLS) init from thread (for parallel init and to avoid blocking the main signaling thread)
• improved: HTTP API URI parameter parsing
• improved: discard old network paths statistics on version change
• improved: queue new call requests
• improved: MWI (voicemail) notifications
• improved: a list of other small changes
• fix: increase timestamp while not sending
• fix: auth for IM SIP MESSAGE
• fix: 10.0.0.0/8 ip addresses detected as any ip
• fix: opus codec payload negotiation
• fix: various other minor bug fixes and improvements

This is a minor new version with a list of bug fixes, improvements and optimizations.

• new: G.722.1 codec
• new: USSD notifications
• new: UUI data (User-to-User Call Control Information as described in RFC 7433)
• new: IMS/3GPP/VoLTE (basic compatibility and features such as USSD or 3GPP SMS. ims3gpp parameter)
• new: RFC 7989 implementation (Session-ID header)
• new: 3PCC (1PCC/3PCC Allow-Events, talk/hold, RFC 3725; enable_3pcc parameter)
• new: streaming from RTP source (previously only raw pcm was supported)
• new: player/recorder usage statistics
• new: auto transport protocol failover attempt on connection failure
• new: support for compact SIP headers
• new: VAD by line
• new: TLS domain validation option (tlspolicy)
• new: possibility to set client certificate (tlsclientcerttype, tlsclientcertfile and tlsclientcertpass parameters)
• new: Alert-Info and Call-Info handling for auto-answer (including Answer-After delay)
• new: dropduplicatepackets parameter to filter out duplicate udp packets
• new: simplified codec settings with the codec and prefcodec parameters
• new: configurable sdp session name (sdpsessionname)
• new: configurable session idle timeout (idletimeut)
• new: possibility to set TOS also for the signaling (sigtos and mediatos parameters)
• new: sip_uui parameter (to specify UUI data for all sessions)
• new: enableautoaccept, autoacceptdelay and autoacceptheader parameters
• new: sipproto parameter (sip, sips, tel)
• new: sendrtpondisabled and sendrtponfailed parameters
• new: sendmediain_conf
• new: maxlinesex parameter
• new: playbacksteamformat parameter
• new: sendmediaout_line parameter
• new: unregextraccounts parameter
• new: checktargetbranch parameter
• new: alloweventsheader parameter
• new: ringtoneout parameter
• new: API_DisableBLF
• new: API_RequestStatus
• new: API_SendSIPMessage
• new: API_SO
• new: API_SendSIP
• new: API_SendSIPMethod
• new: API_SetUUI
• new: API_SetSSLContext
• new: API_SendUSSD
• improved: module loader
• improved: drop packets on first burst long queue
• improved: precahce/preinit winaudio to reduce first packets delay
• improved: API_SetParameters accept also comma as separator instead of \r\n
• improved: optimizations/adjustements for Java 16
• improved: SRTP (strictsrtp)
• improved: TLS transport
• improved: voice recording file upload and other voice recording related fixes
• improved: multiple simultaneous calls session management
• improved: multiple simultaneous calls audio / media lines management
• improved: auto call retry on call failures with changed parameters
• improved: various minor improvements for VAD and AEC
• improved: auto-answer
• improved: file paths and urls normalizations
• improved: SRV records handling for failover or load-balancing servers
• improved: API_StreamSoundBuff accept start parameter 0 (stop the playback) even if buffer is 0
• improved: API_Transfer accept also full SIP URI
• improved: number of simultaneous calls can be up to 512 by default (and this can be increased if needed with the parameter)
• improved: voice record upload on HTTPS
• improved: jitter buffer optimizations
• improved: streaming auto convert to wideband if required
• improved: exception handling
• improved: audio device selection
• improved: sendrtponmuted and rtpkeepaliveival handling
• improved: subnet match calculations
• improved: threads termination on close
• improved: codec - payload number dynamic mapping
• improved: voice records start/stop per line on demand at runtime
• improved: hold, mute reload
• improved: auto-unregister
• improved: local app audio streaming
• improved: register state reports
• improved: SIP signaling processing accuracy and performance
• improved: config file storage and path detection
• improved: guess best line for transfer if multiple calls in progress if line is not explicitly specified
• improved: jitter buffer
• improved: audio packet drop with winaudio delay
• improved: more integration examples
• fixed: srtp key length for AES_CM_256_HMAC_SHA1_80 and AES_CM_256_HMAC_SHA1_32
• fixed: srtp sequence number wraparound
• fixed: incorrect SDP IP in some circumstances
• fixed: don't recreate SRTP object on call hold
• fixed: skip correct wave header for file streaming
• fixed: message size crop with TCP and TLS transport
• fixed: call recording sometime can't store
• fixed: API_StreamSoundStream and API_StreamSoundBuff
• fixed: locale strings handling for certain countries such as Turkish
• fixed: calls to SIP URI (with full URI, not only username part)
• fixed: early media can't be heard
• fixed: stop udp threads after API_Stop
• fixed: free memory after calls
• fixed: wrong call hold sendonly/recvonly indications in some circumstances
• fixed: wrong cseq number with ACK in some circumstances
• fixed: 211 support tickets resolved with existing customers since the previous major release

Quality upgrade with bug fixes, minor improvements and an updated SIP/media stack.

• new: handle multiple A records for the same domain
• new: auto detect other phones on the same LAN
• new: textmessaging parameter
• new: re-register with best IP to help NAT unfriendly servers
• new: backup/failover SIP server (backupserver)
• new: API_Info and infocontenttype to send any SIP INFO message
• new: failback to inband dtmf in SIP INFO not answered
• new: send reregister and reinvite reason
• new: read parameters (settings) also from environment variables (keys must be wp_ prefixed)
• new: try to load callerid/displayname also from Remote-Party-ID
• new: auto redial if bye received within 2 second for some disconnect reasons such as "Bearer capability not available"
• new: auto TCP/TLS failback
• new: change pitch based on audio jitter
• new: contactaddressfailback
• new: normalizenumber, numrewriterules, numrewriterulesadv
• new: configurable dtmfwait
• new: sendmedia_line option
• new: configurable tlsversion
• new: bindlip, localip, bindtolocalip, fawlocalip settings
• new: configurable max simultaneous calls
• new: auto adapt for CPU core count (performance)
• new: auto failback between transport methods
• new: line parameter for voice record
• new: API_GetMySIPURI
• new: API_RTPStat
• new: RTPSTAT and LOG,RTP notifications
• new: API_GetGlobalStatus
• new: GetAccountRegState/API_GetAccountRegStateString
• new: remote party id processing
• new: streaming example
• new: ring tone volume
• improved: command line processing
• improved: call transfer
• improved: DNS resolver
• improved: presence reports
• improved: audio device handling
• improved: VAD (longer silence delay and change limiter)
• improved: AEC
• improved: if no incoming packets, re-invite with changed ip (private -> public or inverse
• improved: STUN handling
• improved: jitter packet drop optimizations
• improved: watch for audio line underrun
• improved: new local ip detection on change
• improved: stun natcheck
• improved: SIP dialog handling
• improved: ports settings
• improved: prestart public IP detection
• improved: actual re-register interval calculation
• improved: headless required display on No X11 DISPLAY java.awt.HeadlessException
• improved: more reliable CHATREPORT messages
• improved: call recording
• improved: audio local streaming
• improved: utf characters: handle cyrillic/unicode characters as the audio device name
• improved: NS service installer progress bar
• improved: better handle dynamic IP changes at runtime
• improved: log to console and log window
• improved: tunnel/direct server failover (when tunnel activated)
• improved: target SIP domain vs IP:port handling
• improved: many other minor improvements
• fix: consider rport in via for changing target URI
• fix: close audio device on release
• fix: TCP/TLS connect to multiple remote peers
• fix: sometime transport reset to UDP
• fix: unregister on exit
• fix: 121 closed tickets from the last release
• fix: many other minor bug fixes

• new: callforwardonnoanswertimeout setting
• new: voicerecording upload retry (also for HTTP uploads)
• new: async refresh audio device list
• new: auto transport protocol detect (udp/tcp/tls) (if not set with the transport parameter)
• new: auto transport udp/tcp/tls failback
• new: auto media encrypt detect (mediaencrypt, SRTP)
• new: chat queue
• new: chat feedback
• new: UPnP support
• new: dtmf support in early media
• new: ability to set the logpath. also accept environment variables like %computername% %username%
• new: ability to disable retry with other codec feature
• new: API_GetLineDetails availability over socket
• new: API_CheckConnection
• new: launch GUI from command line or as lib (API_StartGUI)
• new: handle redial at X-Asterisk-HangupCause: Bearer capability not available
• improved: OPUS codec attributes
• improved: playback/recorder thread polling and prioritizations
• improved: don't send iscomposing if not needed
• improved: group chat
• improved: better handle multiple simulatenous call launch
• improved: GetOtherPartyDisplayName and GetBestDisplayname
• improved: Caller-ID display
• improved: handle logpath = null (will restore default path)
• improved: hasredial allowrecall callretryonreject codecretry
• improved: redial
• improved: offline chat
• improved: username handle with/without domain (full URI or simple username)
• improved: keep alive (auto set by default)
• improved: SRTP handling
• improved: faster startup
• improved: clean shutdown
• improved: mediaench library auto download if missing
• improved: better logs for audio playback thread
• improved: base64 encoder for utf-8 strings
• improved: API_GetStatus for lines
• improved: quick STUN
• improved: TLS transport (keep session on TLS if possible)
• improved: recover from audio error
• improved: STATUS reports
• improved: auto accept call option
• improved: presence state strings (PRESENCE notifications)
• improved: disable fast stun only if improves connectivity
• improved: skip dns lookup on subsequent failure
• improved: dns srv record lookup
• improved: registered state reports
• fix: auth username handling
• fix: transport name in the signaling
• fix: webphone freezing at: EVENT,audioplayer close
• fix: disconnect reason texts

• new: group chat
• new: offline messaging
• new: auto switch codec for conference
• new: capability to change audio format during calls
• new: advanced dial plan rules (dial plan)
• new: handle system environment variables for paths
• improved: main status reports
• improved: local character set handling
• improved: via rport handling
• improved: conference mixer
• improved: presence
• improved: best codec selection
• improved: codec change on reinvite
• improved: dynamic payload handling
• improved: prioritize codec which is already in use by other ep
• improved: conference from wideband call
• improved: audio queue teardown
• improved: cisco CUCM compatibility
• improved: mediaench
• improved: notification handling
• improved: CHATREPORT (with better offline queue handling)
• improved: route and path handling
• improved: sync voice recording for call legs
• improved: best local IP autodetect
• improved: dns lookups
• fix: re-register if no register endpoint
• fix: signaling halt after audio error
• fix: lose connection after a while
• fix: freeze on audio record device disable or change
• fix: catch on GetHostString voip.mizu-voip.com
• fix: direct call to SIP URI
• fix: speex wideband minimum buffer size
• fix: winaudio hung on close
• fix: crash on close
• fix: numerous other minor bug fixes

• new: API_GetNotifications and API_GetNotificationsSync (no need for socket anymore for the notifications)
• new: IMS/3GP/VOLTE support
• new: multi-line remote audio streaming
• new: reports about voice recording (VREC)
• new: mp3 call record option
• new: unsubscribe API
• new: store (rename) previous log file
• new: call with DTMF as: xxx,yyy (first call xxx, then send yyy as dtmf)
• new: sample rate convert
• new: wideband audio streaming (this is for local streams. wideband codec for VoIP calls have been added long time ago)
• new: PLAYREADY notification for local audio streams
• new: API for local streaming from buffer
• new: API_RecFiles
• new: run without audio devices (for example for streaming from/to file or for recording on a headless Linux)
• new: wpapiconnectport setting (set before start if needed)
• new: aectype setting
• new: singleinstance setting
• improved: correct (best match) local bind IP detection
• improved: successive API calls (multiple calls launched at once)
• improved: presence xml handling
• improved: wideband optimizations
• improved: multi-line operations
• improved: call transfer
• improved: non-blocking dns requests in separate thread (with timeouts)
• improved: GetCodecFrameCountTime (to be sent in SDP vs real)
• improved: silence detect alg skip packets during call hold
• improved: GUI flickering
• improved: DeleteOldRecFiles recursive
• improved: DNS record caching with auto refresh
• improved: don't send ping packets if not registered
• improved: chat composing notifications (SIP MESSAGE isComposing)
• improved: applying settings changes vs caching
• improved: improved compatibility with third-party devices (accept malformed SIP packets if doesn't affect security)
• improved: a long list of micro optimizations
• fix: multiple frames per packet for speex
• fix: crash with multiple frames per packet in opus ultra wide-band
• fix: AEC related buffer overrun 
• fix: ftp missed files upload
• fix: special or unicode characters in the audio device name
• fix: CSeq number increase for new REGISTER after old failed
• fix: various SIP signaling related fixes
• fix: 14 other minor bug fixes

• new: headless mode (can be used also on servers with no GUI or X-Window installed)
• new: BLF (support for Busy Lamp Field)
• new: autoprovisioning with opcodes
• new: notification about unregister
• new: wideband conference
• new: in-band DTMF (previously only RFC 2833 and SIP INFO was supported)
• new: API_GetLineDetails to query endpoint details
• new: allowsipuriasusername parameter (for URI input normalization)
• new: AGC for recorded voice and volume normalization for the sides
• new: auto codec switch from wide-band to narrow-band when low quality network is detected
• new: registerex API
• new: OPUS codec pure Java implementation (not need for external library anymore)
• improved: voicemail (display and asterisk compatibility)
• improved: presence
• improved: conference mixer
• improved: multi-line (simultaneous calls)
• improved: PLC (packet loss concealment)
• improved: voice recording quality
• improved: insert multiple custom SIP headers
• improved: minimal built-in user interface (not for the headless version)
• improved: command line parser, config file detection
• improved: example code and documentation
• improved: a long list of other minor improvements
• fix: catch on RefreshDNSCache, catch on HostToIpExFromCache
• fix: InetConnectionTest (when used with public server)
• fix: rejectonbusy
• fix: conference calls with speex
• fix: wideband codec transcoding bug fixes
• fix: special characters in chat messages
• fix: G.729 ptime and framing time fix for voice call recording
• fix: hold/reload sending 0.0.0.0 ipv4 address with the SDP
• fix: 44 other minor issues closed (all the pending issues resolved, no any open ticket left)

• new: presence complete rewrite with support for PUBLISH/SUBSCRIBE/NOTIFY
• new: unicode for chat
• new: public universal API accessible via UDP/TCP/HTTP (clear text/XML/JSON or URL parameters)
• new: ability to receive the media streams from the SDK
• new: multiple simultaneous registrations (multi-account capability via the “extraregisteraccounts” setting)
• new: UPNP NAT (better NAT handling behind UPNP capable routers)
• new: reinvite with private media ip if no incoming audio with stun enabled (handling stun external port mismatch)
• new: more settings such as allowcallredirect, disablesamecall, checkvolumelevel and usecommdevice
• new: chat composing notifications and send via API_SendChatIsComposing
• new: detailed call details (API_GetLastCallDetails)
• new: recorded voice upload via http post (previously only FTP was available)
• new: auto call forward on no answer (“callforwardonnoanswer” settings)
• new: enableaudiostreams setting (set to 0 to disable audio recording and playback)
• improvements: IM delivery robustness
• improvements: SIP stack handle warning and diversion headers
• improvements: stun and turn (earlier public IP discovery, doesn’t use on local LAN’s)
• improvement: various improvements for SIP and mediaench and more native audio features on Windows OS
• improvement: local conference RTP mixer
• improvement: SIP proxy and registrar handling
• improvement: VoIP logs in a separate low-priority thread
• improvement: better proxy detection for UDP/TCP/TLS
• improvement: various improvements for direct encrypted peer to peer media handling
• fix: HTTP API
• fix: config and log file management
• fix: DNS SRV record timeout handling
• fix: AEC for wideband speex and opus
• fix: audio device listing
• fix: reject double outbound calls to same destination
• fix: more than 70 minor fixes and improvements

• new: multiple simultaneous registrations (multi-account capability via the “extraregisteraccounts” setting)
• new: UPNP NAT (better NAT handling behind UPNP capable routers)
• new: reinvite with private media ip if no incoming audio with stun enabled (handling stun external port mismatch)
• new: more settings such as allowcallredirect, disablesamecall, checkvolumelevel and usecommdevice
• new: chat composing notifications and send via API_SendChatIsComposing
• new: detailed call details (API_GetLastCallDetails)
• new: recorded voice upload via http post (previously only FTP was available)
• new: auto call forward on no answer (“callforwardonnoanswer” setting)
• improvements: sip stack handle warning and diversion headers
• improvements: stun and turn (earlier public IP discovery, doesn’t use on local LAN’s)
• improvement: various improvements for mediaench
• improvement: more native audio features on Windows
• improvement: local conference mixer
• improvement: log in a separate low-priority thread
• improvement: http download/upload/autoprovisioning/ssl issues bypass option
• improvement: better proxy detection for UDP/TCP/TLS
• improvement: various improvements for direct encrypted peer to peer media handling
• fix: DNS SRV record timeout handling
• fix: AEC for wideband speex and opus
• fix: audio device list on Windows
• fix: ptime settings for G.729
• fix: reject double outbound calls to same destination
• fix: more than 60 other minor fixes and improvements

• new: voice activity detection, statistics and user notifications
• new: public universal API accessible via UDP/TCP/HTTP (clear text/XML/JSON or URL parameters)
• new: custom skin wizard (internal)
• new: blacklist
• new: voice recording http upload
• improvement: web address in manifest to avoid Java – JavaScript browser popup
• improvement: various tunneling related changes
• fix: SRTP
• fix: tunnel failover to backup server
• fix: select audio device on Windows and MAC

• new: peer to peer encrypted media
• new: presence and contact list management
• new: more flexible configuration
• new: desktop mode
• improvement: SRV DNS failover

• new: stable TLS/SRTP
• new: native audio layer for windows
• new: integrated with the new tunneling module
• new: a new skin template
• improvements: API
• fix: conference and other audio related bugs

• new: opus codec
• new: video module (“as is”)
• new: ogg/vorbis recording
• new: java API
• improvement: aec
• improvement: reregistration
• improvement: rtp read threading
• fix: more than 20 minor bug fixes

• new: skins
• new: recording in gsm format
• new: failowering based on SRV records
• new: voicemail
• new: mediaench (AEC, AGC, denoise) now available also on MAC
• improvement: more than 40 minor improvements and bug fixes
• improvement: JS API
• fix: compatibility with the recent java update
• fix: via branch bug
• fix: other minor issues
• fix: dtmfmode when set to 3 now sends in both formats

• new: denoise filter (noise suppression)
• new: AGC (auto gain control)
• new: the ability to set custom ringtone
• new: call forwarding
• new: silence detection and suppression
• new: streaming audio file to the other end
• improvement: AEC module complete rewrite(currently available on windows only)
• improvement: call recording and sound playback API’s
• improvement: new encryption for TCP and HTTP tunneling
• improvement: API_PlaySound can play both local and remote files
• improvement: HTTP tunneling
• fix: address incomplete bug with INVITE
• fix: one way audio in conference
• fix: some minor issues with codec negotiations

• new: conferencing
• improvement: iLBC codec (now it is enabled by default with low priority)
• improvement: iPhone like skin example and documentation upgrade

• new: voice recording
• new: aec (automatic echo canceller -beta version)
• new: automatic transport detection (failovering from UDP encryption -> TCP-tunneling -> HTTP)
• new: httpsessiontimeout applet parameter
• new: ringtimeout applet parameter
• new: waitforunregister option
• new: option to enable peer to peer calls (direct call to SIP URI)
• new: various new applet parameters and API function to allow more external control
• improvement: disabled nagle algorithm for TCP tunneling
• improvement: auto detect audio card wideband capabilities before the first call is made
• improvement: packet loss concealment enhancements. Plc is enabled by default
• improvement: handling of out of order packets
• improvement: jitter buffer fine-tuning
• improvement: call transfer
• improvement: bigger range for the volume control
• improvement: thread manager
• improvement: random UDP port for tunneling
• improvement: autodetect direct server access possibility (while using tunneling_
• improvement: call duration display in hh:mm:ss format
• fix: always clear old credentials on new settings and on unregister
• fix: API_Dtmf, API_Unregister (waitforunregister)
• fix: iphone skin compatibility with IE, Chrome, Firefox and Safari
• fix: codec change on reinvite

• new: api_playsound
• new: ackforauthrequest
• new: 100rel PRACK support
• new: earlymediasend applet parameter
• new: logtoconsole applet parameter
• new: autoaccept applet parameter
• new: rejectonbusy applet parameter
• new: sendmac applet parameter
• new: call timeout setting
• new: API_GetVersion
• new: API_Chat
• new: API_TransferDialog
• new: iPhone skin
• improvement: show the sip display-name for incoming calls
• improvement: receive chat messages as javascript notifications
• improvement: call to URI (loading the server from URI and not from settings)
• improvement: discparty parameter in CDR records
• improvement: transfertype 3 and 4
• improvement: http tunneling
• improvement: java script event notifications
• fix: background color settings in the chat and audio settings form
• fix: one way audio on some circumstances since version 3.5
• fix: restored compatibility with java 1.4
• fix: authentication with empty parameters (for example empty opaque)

• new: http tunneling using browser http send/receive capabilities to automatically bypass http proxies
• new: plc algorithm (packet loss concealment)
• new: Spanish translation
• new: capability request applet parameter and function call
• new: media timeout option
• new: srv dns record lookup option
• improvement: rtcp option
• fix: final codec offer in ACK caused one way audio problem. Now this can be disabled with the setfinalcodec option.
• fix: sequence number overrun caused media RTP problems after 21 minute

• new: http tunneling
• new: deployment toolkit example
• new: DTMF RFC2833
• new: Toolkit_with_JS.htm deployment example
• new: easy encryption for all applet and java function string parameters
• new: API_SetParameter function
• new: code frame count setting
• improvement: js and jnlp now accept jre 1.4
• improvement: possibility to pass MD5 instead of password
• fix: API_SendDtmf was not able to send more than one DTMF digit at once
• fix: STUN sets external IP even on wrong conditions

• new: accept header
• new: API_MuteEx function
• new: jnlp documentation and examples
• improvement: allow header now list all supported methods
• improvement: remaining credit display timings
• improvement: display sent dtmf digits
• improvement: support for multiple instances
• fix: sdp body is missing from INFO messages sending DTMF
• fix: removed static variables to improve multithread stability

• new: g.729 codec
• new: wideband and ultra-wideband codec
• auto convert mono to stereo sound
• improvement: better audio device handling (try to open all existing device on failure)
• fix: some SIP messages don’t contain the URI in message header

• new: authorization name and display name parameters
• improvement: new discontransfer applet parameter
• improvement: OpenSIPS compatibility
• improvement: new color parameters
• fix: route/record-route handling in transfer, hold and disconnect
• fix: cseq sometime is not increased for subsequent register requests
• fix: ring not stopping on call reject/hangup
• fix: call transfer sends wrong refer-to URI
• fix: display registered status when not registered

• new: more options to enable/disable certain functions
• improvement: call transfer compatibility
• improvement: better handle reinvite requests
• improvement: multiline status management
• improvement: call hold and call transfer
• improvement: changed some default GUI settings
• fix: microphone control change bug
• fix: some incoming calls was dropped because wrong cseq initialization

• new: multiple lines (up to 4)
• new: select audio device
• new: volume controls
• improvement: attended transfer
• improvement: extended authentication options (qop, auth-int)
• fix: call transfer Refer-to URI brackets
• fix: auto dial and register
• fix: handling ACK for 200 OK

• new: call hold option
• new: default volume applet parameters
• new: java to javascript API
• new: javascript to java API
• new: ringtone for incoming and outgoing calls
• improvement: ability to enable/disable/set priority for g711 codec’s
• fix: hold and mute sometimes disabled
• fix: register endpoint timeout
• fix: cdr records not sent to javascript

• new: mute, transfer, redial
• improvement: handling record route
• improvement: rtp packet replay to all request -open NAT
• fix: via branch and to tag was kept persistent across dialogs
• fix: save setting not worked since version 1.6
• fix: redial
• improvement: opening NAT at call begins by sending UDP packets to possible destinations
• improvement: jitter buffer fine-tune and configuration possibilities
• improvement: receiving early media (like ringtones and announcements)

• new: quick STUN
• new: rtport handling
• new: GSM codec
• new: outbound proxy
• new: applet parameters: signalingport, rtpport, register interval
• fix: registration and call timeouts
• fix: authentication sent with all request
• fix: sip stack initialization on Vista
• fix: sip message parser
• improvement: jitter buffer
• improvement: thread priority optimizations

• new: DTMF with INFO
• new: speex codec
• new: IM (chat) with MESSAGE method
• new: basic presence
• fix: sip signaling message handler

• initial stable release (RFC 3261 and 2543, PCMU, PCMA)

• beta version with basic SIP call functionality