WebRTC-SIP Gateway Development

We are constantly improving our WebRTC2SIP stack, introducing new enhancements such as changes to reflect the latest industry standards, new features, bug fixes, optimizations.

All the important modules were developed by Mizutech from scratch with special care for performance, interoperability and robustness. These includes a high performance network stack, full featured SIP and WebRTC stack, media stack, TURN and STUN modules, routing module, rate limiter and many many more. For some modules we are using third-party open source solutions (executable as-is or unmodified linked library binary formats, conform to their license and all laws in all circumstances). This includes openssl (for TLS, HTTPS, WSS), opus (audio codec), speex (audio codec), x264 (video codec), VP8 (video codec), freeswitch (for some PBX features), lego (for Let's Encrypt), slimpftp (for recorded voice file access) and a few others other rarely used tools such as unzip and ffmpeg.

The WebRTC gateway is backward compatible. This means that you can just install over your existing old instance to get the new benefits.
Note: the MRTC gateway consists of multiple modules with different version numbers. (For example the MRTC core, the SIP stack, the WebRTC stack, the media stack, the admin client or the installer itself). These modules are coupled together in the final executable and installed for you with an easy to follow setup wizard.
The below listed version numbers are our main branch numbers and you might see different version numbers for the included software components.

Major changes for the Mizu WebRTC-SIP gateway (MRTC) are listed below.

MRTC Version history

MRTC 2.8 - Wednesday, June 10, 2020

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)

MRTC 2.6.19063 - Friday, June 28, 2019

fix: call authentication bug
fix: download link

MRTC 2.6 - Friday, June 7, 2019

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

MRTC 2.4.2 - Saturday, June 9, 2018

This is an upgrade over the previous v.2.4 major release, containing only bug fixes, improvements and optimizations. No new features were added in this release.

MRTC 2.4.1 - Monday, May 7, 2018

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

MRTC 2.0.2 - Tuesday, June 13, 2017

This is a small upgrade with some bug fixes addressing a few minor issues.

We also fixed the download link on the website (previously the default download link was pointed to an old release).

MRTC 2.0.1 - Saturday, April 22, 2017

This is a new release which can be considered as an upgrade over the previous 2.0 version. Changes includes:

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.

MRTC 2.0 - Wednesday, April 19, 2017

This major new release makes WebRTC-SIP protocol conversion available for everyone without the burden of complex text based configurations, featuring fully automated configuration wizard a GUI based administration.

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)

MRTC 1.4 - Monday, January 30, 2017

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)

MRTC 1.2 - Tuesday, August 2, 2016

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

MRTC 1.1 - Tuesday, June 14, 2016

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

MRTC 1.0 - Sunday, January 17, 2016

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

MRTC 0.8 - Monday, October 5, 2015

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

MRTC 0.2 - Monday, July 27, 2015

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

MRTC 0.0 - Sunday, May 10, 2015

initial research
start working dtls, srtp and webrtc module

More news and software releases can be found here.
Like our facebook page to get notified about new software releases and VoIP industry news.

WebRTC2SIP gateway