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

• new: auto detect websocket type (binary vs text)
• new: auto dtmf send method failover
• new: DTMF triggered call transfer and conference
• new: support for x-forwarded-for
• new: epreg command to list registrations
• new: allowanonymoususername config option
• new: autoaddserverstorouting config option
• new: auto block SIP users on WebRTC only paths
• new: support for Push Notification for SIP (RFC 8599) (push is an optional module)
• new: HTTP/2 support for apple APNS (push is an optional module)
• new: auto block users settings (fail2ban)
• new: auto call transfer failover
• new: support for upper proxy server
• new: recorded calls download/playback from MManage
• new: HA with db failover
• improved: WebRTC stack general improvements
• improved: SIP stack upgraded to latest stable release
• improved: media stack upgraded to latest stable release
• improved: WebRTC to SIP and SIP to WebRTC signaling rewrite changes
• improved: media stack upgraded to latest stable release
• improved: always routed early media by default
• improved: better websocket request detection also from malformed connections
• improved: TCP/TLS transport
• improved: voicerecdownload API
• improved: forward in correct transport protocol to upper servers
• improved: load orig IP for routing
• improved: don't off-route media if voice recording required
• improved: cdr API
• improved: CDR's for WebRTC to WebRTC calls with direct routing
• improved: GetCandidateBestSrflxPort
• improved: TLS proxy thread switch and auto prioritization
• improved: call transfer
• improved: delete old webrtc user list on startup
• improved: dtmf routing
• improved: auto convert dtmf
• improved: caller-id forward
• improved: min-expires forward
• improved: correctly handle UDP/TLS/RTP/SAVPF
• improved: authorization request forward vs answer from local
• improved: get best IP, get if same subnet and get if routable functions
• improved: send our x-call-id back to WebRTC stack
• improved: auto let's encrypt certificate renew
• improved: local WebRTC user authentication
• improved: anonymous username handling
• improved: a long list of minor improvements
• fix: voice recording download sometime not enabled by default
• fix: call fork between SIP and WebRTC
• fix: target URI match target domain for call fork
• fix: to tag handling
• fix: webrtc call-time limiter
• fix: endpoint protocol bug
• fix: sometimes first register not accepted from WebRTC clients
• fix: various transport protocols for call forking
• fix: many minor bug fixes, 44 resolved tickets (all pending issues resolved)

• fix: call authentication bug
• fix: download link

• new: setroutepath setting (use path and route headers. rfc3327)
• new: auto guess best local IP per session (on servers with multiple network interface)
• new: handle latest let's encrypt ACME protocol
• new: stat about upper server registrations (all/success/failed)
• new: remove upper fcm reg if failed to register and auto remove acc with failed reg
• new: resend INVITE on new register (UpdateCallTargetOnNewReg)
• new: call fork across different protocols
• new: allowupperserveronsameip (set automatically)
• new: delaltaddresslist command to delete also addresses from HUsers
• new: cisco cucm upgrade with separate user source address
• new: uniquecontactaddress setting -1: auto (if cisco/cucm detected), 0=no, 1=to sip server only with REGISTER/INVITE, 2=sip to servers, 3=all servers (all eClient endpoints), 4=always (also for client endpoints)
• new: config option to handle: caller is using web browser webRTC, no logins, no registration
• new: mmanage gateway config: handledtmfevent: enable dtmf based features such as transfer or call. 0: no, 1: auto, 2: always (set to 2 if to handle dtmf in gateways)
• new: handle X-AUSR in the SIP signaling
• new: reusedifferentportreg setting (-1: auto (will def to 1,0:no,1:yes))
• new: reuseupperregep setting (-1: auto, 0: no, 1: yes, 2: keep auth)
• new: allowupperserveronsameip setting (false: no, true: yes, upper server is on the same IP like the server)
• new: userautoaddtr setting -1=auto,0=no,1=yes,2=yes and parent must exists auto add traffic sender (usually the same as userautoaddwithowner)
• new: enforceowner setting -1=auto guess, 0=no,1=auto,2=yes. Use 1 for tunnel servers where we provide license for a specific company to avoid too much different free license used by a brand (old enforceuppercompany)
• new: insersdpmid setting 0:no,1:for firefox,2:always
• new: allowforkforwebrtc setting
• new: honormediamuteorhold setting 0=never,1=auto,2=always
• new: refertoforward setting -1: auto, 0: don't forward, 1: auto, 2: forward as-is
• new: forwarddtmf setting 0: no, 1: auto, 2: yes
• new: checksameaddresssdp setting 0: no (old behaviour), 1: yes (default),2=always. set to 1 to route RTP between clients with same public IP but public SDP (because they will not be able to reach each other on their router public IP)
• new: autocreatereguserstrict setting 0=no (def), 1=strict. If autocreatereguser is 1 and autocreatereguserstrict is 1, then don't auto create on INVITE (only on register auth answer)
• new: quick (transport level) answer for OPTIONS requests
• improved: SIP and WebRTC stack updated to latest version with all cumulative changes since 2018 September
• improved: faster DTLS processing
• improved: websocket quick pong packet answers
• improved: websocket WebSocket-Key handling
• improved: on upper server change, handle tb_users fields: Proxyaddress, hwid, ContractComment
• improved: +quick forward only on higher load
• improved: admin client user interface (easier management)
• improved: auto gateway user record handling. auto delete old fcm records
• improved: admin client startup time
• improved: auto codec convert from/to OPUS now will check also server load and OPUS parameters
• improved: roles and features: auto select websocket if webrtc is selected
• improved: forwarding authorization request
• improved: auto remove not important headers on UDP if message is too long (for example a=rtcp-xr)
• improved: TLS protocol (many fixes regarding new TLS connection and TLS related SIP signaling)
• improved: call transfer and call hold
• improved: Autocreatereguser
• improved: check client state on register forward
• improved: delete very old users from maintenance (traffic senders, sip servers, resellers with no childs) if it was auto created
• improved: REBUILDREGCLIENTS should be the same as upper reg ival, so the client endpoints are recreated before old deleted
• improved: rtp routing (set to don't touch but changing c=IP)
• improved: don't forward X-C-Call-ID: especially if same as call-id (from fwd-reg)
• fix: Rtc outbound forbidden from_sip
• fix: multiple upper registration sessions for the same user with different call-id
• fix: wrong displayname bug
• fix: mmanage webrtc check select also stun/turn
• fix: (autosslwildcard) will restart tls proxy within 0 days
• fix: detect upper server ip:port same as local
• fix: StoredProcedures.OpenSP_v_check_pxrules
• fix: Call to itself problems: because traffic sender was not auto created
• fix: fs xml not created at all
• fix: don't delete old reg clients if ep->ep
• fix: don't route calls to sipserver_template 9.8.7.6
• fix: SIP/2.0 403 URI domain not configured locally, relaying forbidden
• fix: tlsproxy doesn't convert url from webportal, so WebApplication->TerminateAndRedirect(WebApplication->ApplicationURL); doesn't work correctly
• fix: tlsproxy https -> http convert issues
• fix: MAXSTRMEMORY is too small
• fix: ERROR,(fast) catch on authprocessdata (5) Access violation at address 00D06CAD
• fix: ERROR, catch id: 126016, lastoop: 1, (msg): Access violation at address 009A7968
• fix: anumberlookup not found
• fix: routing to shared did no match

• new: wildcard certificates (easy TLS configuration also for gateways behind NAT)
• new: push notifications
• new: full OPUS codec support, including call recording and transcoding
• new: automatic settings backup
• new: webrtc ping for keep-alive
• new: call recording for webrtc to webrtc calls (previously only webrtc to sip could be recorded)
• new: allownumbersendbackthis option to remember (last or more) numbers sent back (number,IP) and don't accept the same call back
• new: option in config wizard to announce disconnect reasons (allowdiscmessage)
• new: config wizard: option for offline chat
• new: ability to auto generate local domain name
• new: auto set apiv2md5salt for new servers (at first config or at first start)
• new: config.srtp: 0=disable,1=bypass only,2=use if offered (default),3=force offer,4=reject if not offered
• new: auto shrink logs on low disk space
• new: support for binary websocket (vs text)
• new: SSL/TLS tests
• new: websocket close frame handling
• new: proxyaddress handling in compact mode
• new: cache GetServerConnectAddress
• new: option to turn off registrations (blind accept)
• new: auto disconnect ws tcp connections after some time (especially if server load is high)
• new: voicemail and other SUBSCRIBE/NOTIFY forward (previosly only presence SUBSCRIBE was supported)
• new: detect upper server also from the route header
• new: rewrite sip signaling from WebRTC to make in compatible with all SIP servers
• new: runtime target address rewrite for INVITE when new REGISTER arrives
• new: SSE2 optimizations
• new: built-in and auto generate dtls certificates
• new: support for SQL compact 4.0
• improved: def force incoming calls from SIP to local endusers only (no route back)
• improved: attended transfer
• improved: SUBSCRIBE/NOTIFY routing
• improved: refresh letsencrypt cert 20 days before expire to avoid notification emails
• improved: sipjs compatibility
• improved: conference
• improved: routing between different domains
• improved: video re-invite
• improved: multi-calls
• improved: voice recording forward to http/ftp
• improved: presence
• improved: handling Not Acceptable Here 488
• improved: webrtc to webrtc direct routing (when enabled)
• improved: DTMF forward (convert if necessary)
• improved: auto guess correct called user for incoming calls
• improved: configuration wizard
• improved: auto create user performance
• improved: contact uri handling for sip requests
• improved: contact rewrite
• improved: handling of minimum session expires
• improved: renegotiate codec on hold
• improved: NAT keep-alive packets
• improved: number format handling (ex: tel:+358-555-1234567;postd=pp22 )
• improved: TLS proxy (heartbeat, keep-alive with auto-disconnect, thread polling and handle high number of connections)
• improved: disable cantryudpdirectep if server is on private ip but gateway is on public ip
• improved: quick config wizard: no public ip display; no warning if no domain and no ssl for webrtc gw
• improved: on rtp send on tcp/tls, check queue first (skip if too long) ...or set an initial short send buffer size
• improved: list all apps/listen ports if server cannot bind to sip signaling udp or mainport
• improved: reuse socket SO_REUSEADDR for lego
• improved: optimize opus dll (fast math, etc)
• improved: websocket connection auto-reconnect improvements
• improved: wideband transcoding
• improved: double-nat handling
• improved: auto codec conversion (only when necessary)
• improved: register forward
• improved: handle websocket close packet (close msocket on websocket closed state)
• improved: don't send to websocket if connection not initialized
• improved: country settings
• improved: more tests on windows server 2016
• improved: do not bring up the server role selection even if iscompact iswebrtcgw
• improved: mmanage config wizard company page: default username for admin account: admin
• improved: force lan only sometimes hides the preferred/public ip
• improved: before defaulting to port 80, 5060, check if these ports are already used (by a local iss or sip server)
• improved: improved compatibility with third-party devices (accept malformed SIP packets if doesn't affect security)
• improved: more then 50 other minor improvements for the gateway and the admin client
• fix: caller-id forward
• fix: connectivity check, check domain external ip
• fix: config load bug
• fix: issue when no audio for webrtc to webrtc in firefox
• fix: configuration not saved
• fix: wsuser cache after wsunreg
• fix: incoming webrtc calls bug id 74884
• fix: if bindip is set to all it will display "Invalid IP (All)! Are you sure to continue with this?"
• fix: call to ivr disconnect on hold button press
• fix: speex codec convert
• fix: call transfer REFER and NOTIFY handling
• fix: upper server name missing on second run
• fix: mmanage restart service asking for update record
• fix: mmanage show first steps can't be unchecked
• fix: sip domain not resolved for upper server if only domain name field was set
• fix: the domain you have set doesn't match the external ip
• fix: mmanage analyze doesn't display first start stats
• fix: CandidateRewriteForClientPublic must keep also private IP (just insert public if needed) ....if both clients are behind same LAN
• fix: a few other minor bugs
• fix: 39 minor bug fix

• New: tutorial (step by step setup guide which can be followed even by people with no VoIP knowledge)
• Bug: fixed problem regarding TCP candidates for WebRTC-WebRTC calls (which sometimes caused call disconnects)
• Bug: fixed a few minor annoying issues in the configuration wizard
• Bug: fixed "no config saved" problem which occured in some circumstances with the wizard
• Improvement: TCP candidates now available also for WebRTC to SIP
• Improvements for TURN processing
• A few other improvements for NAT handling
• A few other minor GUI improvements
• Added x64 (64 bit) binaries (important for the Large edition to be able to use more RAM)
• Database back-end options tailored for your exact needs: 
  • Built-In compact (with the Free, Small and Medium editions)
  • SQL Express (with the Medium and Large editions)
  • Full SQL Engine (with the Large and Gold editions)
  Note: For up to 300 simultaneous calls the built-in engine is just fine.
  Although database usage is not that heavy with the default settings (as the gateway will store only configurations and statistics in the database) the SQL engine is more robust then the built-in compact engine and will also increase your gateway performance under heavy traffic, especially above 500 concurrent calls. 

• simplified admin interface but with more webrtc-sip conversion related options
• various bug fixes and improvements for ICE, STUN and TURN negotiations
• TCP media streaming is now available in all circumstances (but the gateway will force or prioritize UDP whenever possible)
• configuration wizard improvement: simplified network configuration
• configuration wizard improvement: auto NAT detect
• configuration wizard improvement: port availability check
• configuration wizard improvement: auto SSL certificate (by Let's Encrypt)
• configuration wizard improvement: auto add upper server as traffic sender and sip server user entry
• various admin GUI improvements
• auto SSL for servers with private IP
• SIP stack performance related improvements
• options for multiple registration/call fork/ring groups
• allow registations to use the routing module (so it can be used with multiple registrars)
• if auto tls is set and local server and not running, make sure that we can bind to 80 and 443, otherwise display warning
• improvements for webrtc gateway turn tcp candidate
• option to route UA to UA calls trough upper server tunnel setting also for webrtc
• reliable and more transparent registrar by forwarding all authentication request to upper server
• dtmf handling improvements (auto detect between RFC 2833 and SIP INFO)
• fix a bug with unregistartions
• fix for webrtc gateway reject registration if incorrect username/password on the upper server
• bug fix for disabled video negotiation
• bug fix to detect real caller user
• bug fix for stun/turn/ice timeout problem
• handle candidate in endpoint (send media there)
• openssl upgrade to latest version
• improvements for chat between sip and webrtc
• call transfer and call forward between webrtc and sip
• implemented tcp candidates
• security: don't accept packets for private ports
• TLS module optimizations
• improvements for best sdp address pickup
• NAT handling improvements, auto optimizations for current network environment
• more media candidates to handle all NAT and firewalls
• a few bug fixes which might affect system stability
• auto not convert sp to ring if ring messages are also received from the same server
• udp relay candidate changes (don't default to port 80 anymore as port 80 UDP is blocked by lot's of ISP by default)
• webrtc-webrtc calls with server side udp and tcp candidate to help bypass any routers
• fixed webrtc voice call recording
• rfc6544 from tcp to udp improvements
• esg_relay_tcp with nagle off and rtp sock like characteristics
• various media routing performance optimizations (real-time routing with minimal delay)

• simplified admin interface but with more webrtc-sip conversion related options
• configuration wizard improvement: simplified network configuration
• configuration wizard improvement: auto NAT detect
• configuration wizard improvement: port availability check
• configuration wizard improvement: auto SSL certificate (by Let's Encrypt)
• configuration wizard improvement: auto add upper server as traffic sender and sip server user entry
• various admin GUI improvements
• auto SSL for servers with private IP
• options for multiple registration/call fork/ring groups
• allow registations to use the routing module (so it can be used with multiple registrars) 
• if auto tls is set and local server and not running, make sure that we can bind to 80 and 443, otherwise display warning
• improvements for webrtc gateway turn tcp candidate
• option to route UA to UA calls trough upper server tunnel setting also for webrtc
• reliable and more transparent registrar by forwarding all authentication request to upper server
• dtmf handling improvements (auto detect between RFC 2833 and SIP INFO)
• fix a bug with unregistartions
• fix for webrtc gateway reject registration if incorrect username/password on the upper server
• bug fix to detect real caller user
• bug fix for stun/turn/ice timeout problem
• handle candidate in endpoint (send media there)
• openssl upgrade to latest version
• improvements for chat between sip and webrtc
• call transfer and call forward between webrtc and sip
• implemented tcp candidates
• security: don't accept packets for private ports
• TLS module optimizations
• improvements for best sdp address pickup
• NAT handling improvements, auto optimizations for current network environment
• more media candidates to handle all NAT and firewalls
• a few bug fixes which might affect system stability
• auto not convert sp to ring if ring messages are also received from the same server
• udp relay candidate changes (don't default to port 80 anymore as port 80 UDP is blocked by lot's of ISP by default)
• webrtc-webrtc calls with server side udp and tcp candidate to help bypass any routers
• fixed webrtc voice call recording
• rfc6544 from tcp to udp improvements
• esg_relay_tcp with nagle off and rtp sock like characteristics
• various media routing performance optimizations (real-time routing with minimal delay)

• auto acquire valid SSL certificate (from Let's Encrypt)
• optimizations to auto detect and sugegst the best network interface to be used (IP:port), both
• globally and per session
• webrtc dtmf rfc2833 implementation
• webrtc call forward 301/302
• insert udp/tcp relay only if caller/called is not on the same lan
• dtmf payload is 126 for webrtc client  ....adjust the server
• forward disconnect reason from upper
• warn if webrtc is enabled but no domain or no ssl cert
• callredcording voicefile headers (now works also without X- prefix)
• auto redial on fail
• fs to fs calls directly without to route to the upper server
• bug fix for signaling ip presented in SDP when running on local LAN
• don't use the current tcp candidate code also for answers from clients (for webrtc-webrtc calls) 
• don't use the localip for webrtc if that is a public IP but not on this box (it will not be able to bind to it)
• detect early media issues. auto reinit webrtc if there is no audio after 3 seconds in the call (or at call connect if already media issue was detected)
• “inbounddtmf” and “outbounddtmf”  config options
• if no rfc 2833 in SDP, disable 2833 dtmf and switch to INFO 
• settings in config wizard for maxlineforautotunnelusers, maxlinefornewunknownparents 
• separate x86 and x64 modules
• improvements for custom SIP headers
• user online/offline check improvements
• fix wrong sdp bug 18874
• ring config
• webrtc conferencing
• add presence support
• ringback improvements
• config wizard auto add special users
• set pcmu/pcma codec only if recording needed
• fix "max calls reached" issues
• include vc++ runtime into the installer
• mmanage "how to connect" details for endusers
• turn module improvements
• tlsproxy autodetect listenipex (like inconfig wizard). also pass listenipex from config   
• auto codec convert on 488 Not acceptable here
• fix for sending ACK to 200 OK with wrong via branch and wrong to tag
• rtpwritefirst only for private ip's, not to servers
• add max call time and ring time options
• webcallme implementation

• fix webrtc gw incoming call ring issue
• fix webrtc gw double ring
• fix 100% cpu issue
• webrtc reconnect from both the disconnect and error events
• webrtc gw stun port mainaportudp
• tcp/udp turn with insertservercandidate=1 with firewall blocking between client devices
• insert our own candidate and forward the media
• fs_directroutrfstofs option
• implement fast path between local users
• dial groups
• auto enable on firewall
• release security restriction if debug is set
• restrict enduser/internal calls
• def incoming/public calls (allow only from local ip)
• external webrtc supervisor module
• extract display name from webrtc; force the webrtc to forward also the caller displayname
• auto nat handling review
• optimize for fast lookup (cache all after mainlogic fully started) 
• auto detect nat ip handling from source address; auto nat handling allow ip (check from
• where it is registering); autodetect if rtp routing is required (auto call a public ivr and check incoming rtp)
• detect if the 2 peers are behind the same NAT and don't route media in this case (if they have the same from IP)  
• add options for turn and stun (protocols vs features)
• fix webrtc server don't allow calls if previously registered with wrong username/password 
• allow udp only for webrtc to webrtc calls if fs_directroute is 3
• turn over udp (for peer to peer calls)
• implement chat and call recording

• implemented built-in turn and stun modules (rfc 5766, rfc 5389)
• add RTMP protocol module
• chat between sip and webrtc
• auto acquire valid domain
• auto delete old logs
• webrtc calls routing improvements
• bind ip related fixes
• add video support
• implement custom SIP headers
• catch simple websocket communication and change on the fly if needed
• special extensions (redial last called/caller, say time)
• buf fix when callerid is different from username
• handle opus codec
• fix webrtc chat not working: webrtc to SIP, webrtc to webrtc
• fix no audio related issues
• fix dtls related issues
• bypass/forward upper or thru registration
• increase webrtc priority if address is set

• add tls-websocket forward for webrtc
• more call options between WebRTC and SIP
• flexible routing implementation
• GUI user interface for webrtc-sip gateway
• quick answer stun packet
• wsuser password encryption
• fix invalid password issue
• add external IP to SDP on webrtc
• webrtc crash fix
• webrtc one side audio fix
• tls parameter
• fix missing dll issues
• high performance websocket improvements
• implement auto self-configuration

• initial release
• websocket support
• SIP stack integration
• RTP stack integration
• WebRTC stack integration
• DTLS integration
• SRTP integration
• sip message between 2 webrtc client
• basic WebRTC-SIP calls

• initial research
• start working dtls, srtp and webrtc module