The MRTC gateway is intended to provide the most effective and reliable solution for WebRTC to SIP protocol conversion.

• All-in-one: The webrtc2sip gateway includes everything needed for successful and reliable webrtc-sip conversion with built-in TURN and STUN modules, auto generate valid TLS certificate, DTLS/SRTP encoder/decoder, codec conversion, flexible routing, conversion between WebRTC and SIP for voice, video, chat, call transfer and many other features.
• Easy to use configuration wizard: Correct and optimal WebRTC-SIP configuration can be a real challenge. This is not the case with the mizu gateway as by providing a few details (such as your IP) on its GUI configuration wizard, it will auto-optimize itself for your use-case, including NAT configuration, TLS certificate management, port settings and TURN/STUN settings optimized for your hardware and network.
• Works in all network conditions: The webrtc gateway will automatically adapt to your network/NAT and auto-decide the best transport method for all sessions, thus providing full compatibility for all browsers regardless of the client side NAT or firewall (will try to use UDP for media by default -direct p2p or via relay- and if necessary it will fail-back using TCP port 80 which is enabled on most firewalls). The media capabilities are also auto-negotiated and the "best" available codec is always selected by default (including auto codec conversion when required).
• Compatibility: The webrtc gateway is based on open standards, compatible with all SIP servers/PBX/softswitch (such as Asterisk, Cisco, Freeswitch, 3CX, FreePBX, Elastix, OpenSIPS and many others), all SIP endpoints (IP Phones, gateways, ATA's and softphones such as X-Lite), all OS and all browsers (such as Chrome, Firefox, Opera, Edge or via plugins in browsers where WebRTC is still not built-in such as IE and Safari).
• Reliability: Some IPPBX and softswitch has built-in webrtc module. However, most of the available implementations will offer you inferior quality which is not suitable for commercial deployments. For example, most of the available webrtc stacks will work only when used from simple networks and will not work from corporate networks where more rigorous NAT and firewall rules are applied (such as allowing only TCP 80 and 443 for HTTP/HTTPS traffic).
  We take special care to make sure that our solution offers high availability and the best possible call quality in all network conditions regardless of client/server environment.

• GUI: Easy setup and configuration wizard and GUI based management including routing configuration, statistics, CDR's, system monitoring, detailed logs
• Core features: voice/video calls, dtmf, chat, presence, forward, transfer between WebRTC and SIP
• UC: unified communication and social networking capabilities for easy call between users with support for voice, video, file transfer, chat, presence and screen sharing 
• Extra features such as chat and call recording, call queue, call fork, call waiting, flexible routing, number rewrite rules, speed dial, SMS termination, IVR, API access and many others
• Media quality: auto-detect the "best" available codec to use and transport method, apply QoS rules, minimize media path length and delay and enable direct peer-to-peer media whenever possible
• Optimized: automatic intelligent offloading the media routing whenever possible (for example for WebRTC to WebRTC calls the media can usually bypass the server)
• Flexible: while the gateway will load optimized settings automatically by default, you have hundreds of options to fine-tune the details if you have any special requirements
• Robust: auto reconnect, renegotiate, reroute, call-fork, offline chat, self-monitoring with health checks and heartbeat
• High availability: HA is ensured with client and/or server side load balancing and automatic failover. Possibility to run multiple instances as active-active or master-slave. Multiple route sets are also supported.
• Support: we are developers here with a vast experience in VoIP, capable of handling your SIP-WebRTC related requests and issues/emergencies in a professional manner

Protocols:

• Transport protocols: UDP, TCP, TLS, DTLS, HTTP/HTTPS, WebSocket, secure WebSocket
• Call protocols: WebRTC, SIP, IMS, RTP/RTCP, ICE, STUN, TURN, DTLS, SRTP
• Supported protocol conversions: WebRTC to SIP, SIP to WebRTC, WebRTC to WebRTC, SIP to SIP
• Codecs: G.711 (PCMU and PCMA), Opus, Speex, G.722, G.729, G.723, iLBC, GSM, VP8, VP9, H.264, H.265
• Automatic codec conversion when necessary (or force manually)
• NAT and firewall traversal using auto-detected transport methods (UDP or TCP), optimal port availability, ICE, STUN, TURN and advanced media address negotiation capabilities
• Works as WebRTC server, SIP proxy or B2B gateway (UAS/UAC)
• VoIP Push Notifications (Both Google FCM and Apple APNS/PushKit are supported)
• Security: auto acquire Let's Encrypt certificate for HTTPS/WSS/DTLS/SIPS, border security, media security using SRTP, exchange of security keys using DTLS and SDES, rate-limiting, dos attack prevention, topology hiding (B2BUA) and many others
• SIP (RFC 3261) and related RFC's. WebSocket Protocol as a Transport for SIP (RFC 7118). Push Notification for SIP (RFC 8599).

Compatibility:

• On the server side it is compatible with all PBX/VoIP server/SIP trunk/proxy/gateway/carrier supporting the SIP protocol such as Asterisk, 3CX, Broadsoft, Brekeke, Yate, FreePBX, Elastix, Trixbox, Voipswitch, FreeSWITCH, Cisco, Siemens, Huawei, NEC, Mitel and others.
  Multiple SIP server support (route calls to one ore more SIP server, accept calls from one ore more sip servers).
  Direct SIP peers are also supported. 
• On the client side you can use any library implementing WebRTC and SIP over WebSocket as specified in RFC 7118, compatible with WebRTC stacks present in browsers like Chrome, Firefox, Edge, Opera and others, WebRTC plugins for IE or Safari or native libraries such as PJSIP.
  All the common WebRTC SIP clients and JavaScript WebRTC libraries are supported such as Mizu WebRTC SIP client, SIPML5, JSSIP, SIP.JS and others. 
  Works from smartphone, tablet or desktop, using any operating system (Windows, Linux, MAC, Android, iOS).
• Beside browser to SIP and SIP to browser, the gateway also has full support for browser to browser calls and SIP to SIP calls.

• the WebRTC protocol suite: websocket (WS/WSS) for signaling, TURN/STUN/RTP candidates for ICE and DTLS/SRTP for media (usually used by endusers from their browsers)
• and the SIP protocol suite: SIP signaling over UDP/TCP, RTP/RTCP for the media and support for various SIP extensions (fully compatible with your existing IP-PBS or softswitch)

The MRTC (Mizutech WebRTC to SIP gateway) is an “all-in-one” solution for WebRTC / SIP protocol conversion with all the necessary modules built-in and with great care for the details such as various connectivity options for all network conditions, providing a reliable service for your users.   

• The WebRTC2SIP gateway acts like an SBC between the WebRTC clients and your SIP server offering various services such as registrar, routing, proxy or B2BUA, rtcweb breaker, ICE and media transcoder. It will work transparently, so there is no need to change any settings for your exiting SIP server to handle the WebRTC traffic. It will forward all authentications to your softswitch in a completely transparent way using common SIP digest authentication, so there is no need for any user management on the gateway.
• Once the WebRTC client is started, it will connect to the gateway via a WebSocket connection and will start to register. These Websocket packets coming in HTTP/TCP are then converted by the gateway to plain SIP signaling and forwarded to your SIP server (usually by UDP but you can set also TCP for the SIP transport).
• Before or at call connect the RTC clients are performing ICE lookups to gather its own and the peer media addresses (transport:IP:port combinations where the media needs to be sent). The WebRTC gateway will handle and answer these STUN and TURN request in an intelligent way, which will result in an optimal media path.
• With the call setup (INVITE) the WebRTC client will send the above collected addresses as ICE candidates in SDP. The gateway will collect them and will also add a few extra candidates (UDP and TCP relay via the gateway) which can be used when no direct path has been found between parties or DTLS/SRTP needs to be converted to RTP.
• At call setup the gateway will negotiate the best possible media parameters based on circumstances (client capabilities, client bandwidth, server capabilities, server load, configurations and other factors).
• While in call, the WebRTC gateway will convert the DTLS/SRTP media from WebRTC (which is usually streamed in UDP but sometime in TCP) to plain RTP/RTCP which can be handled by your SIP server (Softswitch, IP-PBX, proxy or other equipment).
• If necessary (when no common codec found during media negotiation), it will convert also between WebRTC codec (such as G.711 or OPUS) to common codecs used in telecommunication (such as G.729 or G.723) performing transcoding such as OPUS to G.729 or G.729 to G.711. 
• The WebRTC gateway will also handle extra features such as dtmf, call forward, call transfer, call fork, conference, chat, SMS, video, file transfer, presence and many others. Some of these features can be directly mapped from WebRTC to SIP with protocol conversion, others might need unique handling and capability negotiation with the client software.
• By default the gateway will be configured with optimal defaults based on the basic settings you provide (IP, NAT, etc), however most parameters can be also modified/configured manually, such as using different ports, forcing a specific DTMF mode or forcing a specific codec conversion. 

Use-case:

Add WebRTC and call from browser capabilities for any SIP server allowing the usage of all popular web sip clients such as webphone, sipml5, sipjs, jssip, jscommunicator and many more.