WebRTC-SIP Gateway Quick Start
<!-- If the browser does not support frames put the first topic content here --> <div class="WordSection1"> <div style='border:none;border-bottom:solid #4F81BD 1.0pt;padding:0in 0in 4.0pt 0in'> <p class="MsoTitleCxSpFirst" align="center" style='text-align:center'><span style='font-size:36.0pt'>MRTC WebRTC-SIP Gateway</span></p> <p class="MsoTitleCxSpLast" align="center" style='text-align:center'><span style='font-size:18.0pt'>Quick Start Guide</span></p> </div> <p class="MsoNormal"> </p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The Mizu </span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP.aspx"><span style='font-size:14.0pt;line-height:115%'>WebRTC-SIP Gateway</span></a><span style='font-size:14.0pt;line-height:115%'> (MRTC) is a full stack protocol converter between WebRTC and SIP, including all the modules needed for optimal signaling and media conversion (ICE, TURN and STUN are built-in). Using this software you can initiate and receive calls with WebRTC clients (usually running in browsers) via your existing SIP server.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>MRTC can be installed on any Windows OS and it runs as a Windows service (NT service).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You should install it on a Server or PC close to your existing SIP server (Softswitch or IP-PBX). It can be run also from a virtual machine. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For up to 100 simultaneous calls any PC is fine which can barely run Windows, such as a dual core Xeon with 4 GB RAM and 30 GB free disk space. See the </span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP/Prerequisites_for_WebRTC_SIP_gateway.aspx"><span style='font-size:14.0pt;line-height:115%'>requirements</span></a><span style='font-size:14.0pt;line-height:115%'> if you have more traffic.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><u><span style='font-size:14.0pt;line-height:115%'>Follow these steps to get started:</span></u></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>1)<span style='font:7.0pt "Times New Roman"'> </span></span></b><b><span style='font-size:14.0pt;line-height:115%'>Download</span></b><span style='font-size:14.0pt;line-height:115%'> the MRTC installer from </span><a href="https://www.mizu-voip.com/G/download/MRTC_Setup.exe"><span style='font-size:14.0pt;line-height:115%'>here</span></a><span style='font-size:14.0pt;line-height:115%'> (this is the free version for up to 20 users and 5 simultaneous calls)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>2)<span style='font:7.0pt "Times New Roman"'> </span></span></b><span style='font-size:14.0pt;line-height:115%'>Double-click to start the <b>install</b> process and follow the instructions (requires Administrator rights)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>3)<span style='font:7.0pt "Times New Roman"'> </span></span></b><span style='font-size:14.0pt;line-height:115%'>Follow the <b>Configuration Wizard</b>:</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Once the install completes, it should automatically start the MManage admin client with its Configuration Wizard. Otherwise launch the “MManage.exe” application and go to “Config” menu -> “Configuration Wizard”:</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Follow the “Quick/Auto configurations” Wizard type for the easiest setup</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Take care of the “Bind IP” and “Public IP” settings if your server has multiple networks or you are behind NAT</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Set a domain name and select the “Auto SSL” checkbox if you need secure websocket (WSS). WebRTC clients from Chrome browsers will not work if you don’t host your webpage on HTTPS or you MRTC gateway don’t use WSS.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>4.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>If you haven’t set a domain or SSL, then you might configure your browser or app to allow also unsecure websocket (WS) for WebRTC</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>5.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Set the upper server to your existing SIP server address (also set the :port if your server is not using the standard 5060 UDP port)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>6.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Click “Next” and “Apply” to save the settings</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>4)<span style='font:7.0pt "Times New Roman"'> </span></span></b><b><span style='font-size:14.0pt;line-height:115%'>Note</span></b><span style='font-size:14.0pt;line-height:115%'>: </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>You don’t need to change any settings on your existing SIP server</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>You don’t need to manage users/extensions on the Gateway (manage them on your SIP server as you did it before)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>The gateway can be also used with more than one SIP server. See the </span><a href="https://www.mizu-voip.com/Portals/0/Files/WebRTC_SIP_Gateway_Doc.pdf"><span style='font-size:14.0pt;line-height:115%'>Guide</span></a><span style='font-size:14.0pt;line-height:115%'> if you wish to use one gateway with multiple SIP servers.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>There is no any maintenance required by the MRTC gateway. Once configured properly and started, it will run forever without the need for any maintenance work as it will self-manage itself (including deleting old logs and auto-adapting to environment and network conditions)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>5)<span style='font:7.0pt "Times New Roman"'> </span></span></b><b><span style='font-size:14.0pt;line-height:115%'>Ports / NAT / Firewall: </span></b></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Enable (on your external firewall) and forward (on your router/NAT device if you are behind NAT) the required websocket/SIP/RTP/etc ports. These can be listed from Config menu -> Network -> Active ports. </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Make sure that these ports are not used by some other application such as a local IIS web server (in this case either change the MRTC ports or the third party app port or bind them to separate IP address). </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>If your WebRTC-SIP gateway is behind NAT, auto SSL will work only if you forward ports 80 and 443 on your router from the internet (These ports are required to acquire a “Let’s Encrypt” certificate and can’t be changed to other non-standard ports).</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Enable the services executables on your Windows firewall.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>6)<span style='font:7.0pt "Times New Roman"'> </span></span></b><b><span style='font-size:14.0pt;line-height:115%'>Start the gateway</span></b><span style='font-size:14.0pt;line-height:115%'> service if not already started. This can be done from:</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>MManage -> Control menu -> Start</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>or from Windows Service manager (services.msc) -> “mserver” entry (right click and select “Start”)</span></p> <p class="MsoNormal" style='margin-top:0in;margin-right:0in;margin-bottom:0in; margin-left:.25in;margin-bottom:.0001pt'><span style='font-size:14.0pt; line-height:115%'>Your WebRTC-SIP proxy is ready to accept connections and calls at this stage.</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>7)<span style='font:7.0pt "Times New Roman"'> </span></span></b><b><span style='font-size:14.0pt;line-height:115%'>Configure your WebRTC client</span></b><span style='font-size:14.0pt;line-height:115%'>: </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>To find out how to configure your WebRTC client, go to <b>“Help” menu -> “Client configuration”.</b> This will display clear and easy to follow instructions about how exactly you will have to configure your WebRTC client app.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>8)<span style='font:7.0pt "Times New Roman"'> </span></span></b><b><span style='font-size:14.0pt;line-height:115%'>First quick test call</span></b><span style='font-size:14.0pt;line-height:115%'>:</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Open your browser with your favorite WebRTC client, enter the settings from the above mentioned settings with SIP account A (username/password valid on your SIP server) and connect/register to your SIP server via the MRTC gateway</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Open your favorite SIP client and register directly to your SIP server with account/extension B</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;color:#984806'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Make calls between A and B</span></p> <p class="MsoNormal" style='margin-top:0in;margin-right:0in;margin-bottom:0in; margin-left:.25in;margin-bottom:.0001pt'><span style='font-size:14.0pt; line-height:115%'>Note: </span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Alternatively you make also WebRTC to WebRTC calls or WebRTC -> PSTN calls to landline/mobile phone numbers if your SIP servers allows outbound calls</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Calls to outbound PSTN/carrier/SIP trunk/mobile/landline will be handled in the exact same way as any simple WebRTC->SIP call</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>You can also make WebRTC to WebRTC calls (both endpoint running from browsers with SIP credentials valid on your softswitch/IP-PBX)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><b><span style='font-size:12.0pt;line-height:115%;color:#984806'>9)<span style='font:7.0pt "Times New Roman"'> </span></span></b><b><span style='font-size:14.0pt;line-height:115%'>More</span></b><span style='font-size:14.0pt;line-height:115%'>:</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Following the above steps should fulfill most of your needs as the gateway will auto configure and fine-tune all its modules for optimal WebRTC-SIP protocol conversion out of the box. However there are a lot more you can do with the gateway such as using it with multiple SIP servers, optimize PBX features, run health analysis, setup VoIP push notifications, export CDR records, set call recording, handle special NAT requirements or optimize (avoid) codec transcoding.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For more advanced needs, you can:</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Re-run the configuration wizard in “Detailed wizard” mode </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Change any global settings from the “Configuration” form </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Change per user settings from the “Users and devices” form</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Work with multiple SIP servers (configure as “SIP Server” and “Traffic sender” on the “Users and devices” form and add to your dial plan from the “Routing” form)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check the other built-in modules (open various forms in MManage)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.55in;margin-bottom:.0001pt;text-indent:-.3in'><span style='line-height:115%;font-family:Symbol;color:#984806'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check the </span><a href="https://www.mizu-voip.com/Portals/0/Files/WebRTC_SIP_Gateway_Doc.pdf"><span style='font-size:14.0pt;line-height:115%'>Guide</span></a></p> <div style='border:none;border-bottom:solid windowtext 1.0pt;padding:0in 0in 1.0pt 0in; margin-left:.25in;margin-right:0in'> <p class="MsoListParagraphCxSpMiddle" style='margin:0in;margin-bottom:.0001pt; border:none;padding:0in'><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoListParagraphCxSpLast" style='margin:0in;margin-bottom:.0001pt; border:none;padding:0in'><span style='font-size:14.0pt;line-height:115%'> </span></p> </div> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span class="MsoHyperlink"><span style='font-size:14.0pt; font-family:Symbol;color:windowtext;text-decoration:none'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP.aspx"><span style='font-size:14.0pt'>WebRTC-SIP Gateway home page</span></a></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><a href="https://www.mizu-voip.com/Portals/0/Files/WebRTC_SIP_Gateway_Doc.pdf"><span style='font-size:14.0pt'>Documentation</span></a></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP/WebRTC-SIP-Gateway-Pricing.aspx"><span style='font-size:14.0pt'>Licensing</span></a></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><a href="mailto:info@mizu-voip.com?subject=WebRTC-SIP%20Gateway"><span style='font-size:14.0pt'>Contact</span></a></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;line-height:normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Copyright © </span><span style='font-size:14.0pt;line-height:115%'>Mizutech SRL</span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.25in;margin-bottom:.0001pt'><span style='font-size:14.0pt; line-height:115%'> </span></p> </div>
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WebRTC-SIP Gateway
