WebRTC-SIP Gateway
<!-- 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='position:relative;z-index:251658240;left:-5px;top:-4px;width:839px; height:1070px'><img width="839" height="1070" src="WebRTC_SIP_Gateway_Doc_files/image001.png"></span><span style='font-size: 36.0pt'> </span></p> <br clear="ALL"> <p class="MsoTitleCxSpMiddle" align="center" style='text-align:center'><span style='font-size:36.0pt'>MRTC Documentation</span></p> <p class="MsoTitleCxSpLast" align="center" style='text-align:center'><span style='font-size:18.0pt'>Mizutech WebRTC-SIP gateway User Guide</span></p> </div> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoTocHeading">Contents</p> <p class="MsoNormal"> </p> <p class="MsoToc1"><a href="start.htm#_Toc478373663">About<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>3</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373664">Features<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>3</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373665">How It Works?<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>3</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373666">Legal Agreement<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>4</span></a></p> <p class="MsoToc1"><a href="start.htm#_Toc478373667">Setup and configuration<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>4</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373668">Install<span style='color:windowtext; display:none;text-decoration:none'> </span><span style='color:windowtext;display:none;text-decoration:none'>4</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373669">Configuration Wizard<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>5</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373670">SIP Server Configuration<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>6</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373671">WebRTC Client Configuration<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>6</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373672">Troubleshooting<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>7</span></a></p> <p class="MsoToc1"><a href="start.htm#_Toc478373673">Management and Monitoring<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>8</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373674">MizuManage<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>9</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373675">User Management<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>10</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373676">Add SIP Server<span style='color: windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>11</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373677">Add User<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>12</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373678">Outbound Routing<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>12</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373679">Inbound Routing<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>13</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373680">CDR<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>14</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373681">Statistics and Monitoring<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>14</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373682">Start/Stop<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>15</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc478373683">Backup/Restore<span style='color: windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>15</span></a></p> <p class="MsoToc1"><a href="start.htm#_Toc478373684">Additional Settings<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>15</span></a></p> <p class="MsoToc1"><a href="start.htm#_Toc478373711">Resources<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>38</span></a></p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <p class="MsoNormal"> </p> <h1 style='margin-top:0in'><a name="_Toc478373663"><span style='font-size:16.0pt; line-height:115%;font-family:"Calibri",sans-serif'>About</span></a></h1> <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 Gateway</span></a><span style='font-size:14.0pt;line-height:115%'> (MRTC) is a software solution to convert the WebRTC protocol family to the SIP protocol family.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The MRTC software runs as an NT service on Windows operating system and includes all modules for efficient protocol conversion between WebRTC and SIP.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373664"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Features</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'> </p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The most important features are listed below:</span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>Full stack conversion between WebRTC and SIP</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>GUI based management with real-time monitoring and detailed statistics</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Multiple SIP server support for both outbound and inbound</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Convert Websocket (WS/WSS) to plain UDP SIP signaling</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Convert WebRTC media (DTLS/SRTP) to plain RTP/RTCP over UDP</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Media path negotiation with ICE, TURN, STUN</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Optimal media and codec negotiation between WebRTC and SIP</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Codec transcoding when necessary (handled automatically or can be forced)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Supported <span style='color:black'>codecs</span>: G.711, G.729, G.723, GSM, iLBC, OPUS, Speex, G.722, VP8, H.264 </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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Flexible configuration and routing rules</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Auto TLS certificate</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Push notifications (for Web, Android and iOS)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Intelligent NAT/firewall bypass using both UDP and TCP candidates and shortest path detection</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Security: border security, media security, rate-limiter, DOS attack protection, topology hiding</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Support for a broad range of VoIP features including voice, video and unified communication </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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>PBX/Class 5 features: DTMF (RFC 2833 and SIP INFO), call transfer, call forward, mute, hold, re-invite, caller-id, call fork</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Extra features: call rerouting, voice recording, IM recording, file transfer, conference, voice-mail, presence, chat, SMS, IVR, P2P, callback, webcall, click-to-call and many others</span></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> <h3 style='margin-top:0in'><a name="_Toc478373665"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>How It Works</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The WebRTC-SIP gateway acts as a relay between the WebRTC clients (usually browsers) and your SIP server(s) (IP PBX, Softswitch, SIP proxy or other SIP capable equipment). MRTC includes all the necessary modules for optimal protocol conversion regardless of your WebRTC or SIP software and network circumstances. The gateway will transparently convert the traffic from the WebRTC clients to plain SIP/RTP. No need for any changes or re-configuration on your existing server.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'><img border="0" width="726" height="146" id="Picture 1" src="WebRTC_SIP_Gateway_Doc_files/image002.png"></span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>More details about how it works can be found on the </span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP.aspx#dnn_ctr1708_dnnTITLE_titleLabel"><span style='font-size:14.0pt'>MRTC homepage</span></a><span style='font-size:14.0pt'>. General technical details about WebRTC-SIP gateways can be found </span><a href="https://www.mizu-voip.com/Support/Wiki/tabid/99/Default.aspx?topic=WebRTC-SIP+gateway+definition"><span style='font-size:14.0pt'>here</span></a><span style='font-size:14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373666"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Legal Agreement</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Mizu®, MRTC, webphone, Mizu WebRTC-SIP gateway, Mizu WebRTC-SIP proxy the names associated with Mizutech products are trademarks and/or service marks of Mizutech SRL. All other trademarks are property of their respective owners. This product may contain open source software freely downloadable from Mizutech website.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>This SOFTWARE is licensed, not sold. All rights reserved for MizuTech SRL.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>By installing, copying, or otherwise using this product, you acknowledge that you have read, understood, and agreed to be bound by the terms and conditions of the End User License Agreement for this product.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The software is provided "as is" without any warranty of any kind.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Mizutech makes no warranties, liability or responsibility for the accuracy of this document and makes no representations about the suitability of the information contained in this document for any purpose. In no event shall Mizutech be liable for any direct, consequential, incidental, special, punitive or other damages caused by this document or the MRTC software.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can order and purchase a license for this software from </span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP/WebRTC-SIP-Gateway-Pricing.aspx"><span style='font-size:14.0pt'>here</span></a><span style='font-size:14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Contact us with any questions at </span><a href="mailto:info@mizu-voip.com?subject=WebRTC-SIP%20Gateway"><span style='font-size:14.0pt'>info@mizu-voip.com</span></a><span style='font-size: 14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h1 style='margin-top:0in'><a name="_Toc478373667"><span style='font-size:16.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Setup and configuration</span></a></h1> <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'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'>This is the most important chapter in this Guide.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'> </span></i></p> <h3 style='margin-top:0in'><a name="_Toc478373668"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Install</span></a></h3> <p class="MsoNormalCxSpFirst" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:black'> </span></p> <p class="MsoNormalCxSpMiddle" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:black'>The MRTC gateway has a standard easy to use installer.</span></p> <p class="MsoNormalCxSpMiddle" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:black'>The gateway shall be installed on a Windows machine (server or PC) somewhere near your softswitch OR near your customers OR in the path between (to avoid long media path).</span></p> <p class="MsoNormalCxSpLast" 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'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Make sure that your system meets 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%'>minimum requirements</span></a><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'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Download 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%'> </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Double-click to start the install 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'><span style='font-size:14.0pt;line-height:115%'>4.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Once the install completes, it should automatically start the Admin client with its Configuration Wizard. This is discussed in the next chapter.</span></p> <p class="MsoListParagraphCxSpLast" 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> <h3 style='margin-top:0in'><a name="_Toc478373669"></a><a name="_Toc285322898"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Configuration</span></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'> Wizard</span></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For the basic server configuration you should walk through the detailed <b>configuration wizard</b> which is started automatically at first usage or accessible from the “Tools” menu -> “Configure”.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Don’t change any setting that you don’t fully understand, just click on the “Next” button in this case. Most of the settings are self-explanatory with a short description near them and/or a hint if you hold the mouse over a control.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Pay <b>special attention to the “Network” and “SIP server”</b> pages:</span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>on the Network page enter the correct IP and NAT configuration ("offer services for")</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>make sure that the ports used by the WebRTC-SIP proxy (SIP port, Access port, Secure port) are not used by some other application such as a local 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:.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><span style='font-size:14.0pt'>you need a domain and SSL certificate, otherwise it won’t work in Chrome (these can be obtained automatically)</span></p> <p class="MsoListParagraphCxSpLast" 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><span style='font-size:14.0pt'>on the SIP server page enter your PBX/Softswitch details (later you can add </span><a href="start.htm#_Add_SIP_Server"><span style='font-size:14.0pt'>more</span></a><span style='font-size:14.0pt'> SIP servers if needed)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Click “Apply” on the last page to save the configuration. Check the displayed messages and instructions.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>If you select the “(Re)Start the server” checkbox, then the gateway service should start automatically. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Otherwise start it from the “Control” menu or use the Windows Service manager and start the “mserver” service.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'><br> When you start the service for the first time, it is a good idea to open the “Analyze” form to check for any potential issues. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Once the service is started, you can connect with WebRTC clients and make calls.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Later you can add more servers or modify advanced configuration </span><a href="start.htm#_Additional_Settings"><span style='font-size:14.0pt'>options</span></a><span style='font-size:14.0pt'> after your needs.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'>Note: for WebRTC to work in Chrome (and other browsers using the Chrome engine such as Opera), you need to host your webpage on HTTPS (secure HTTP with valid SSL certificate) and you must use WSS (secure Websocket) for the WebRTC proxy connection. This means that you must specify a (sub)domain name and enable auto-ssl or set your own certificate for the gateway.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'> </span></i></p> <h3 style='margin-top:0in'><a name="_Toc285322899"></a><a name="_Toc478373670"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>SIP Server Configuration</span></a></h3> <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%'>A default/first SIP server record is <b>automatically created</b> for you once you complete the Configuration Wizard.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Use the “Add Upper SIP server” form (available from “Tools” menu -> “Add or import users”) <b>in case if you have more SIP servers</b> where the traffic from the clients needs to be sent (WebRTC -> SIP) or accepted from (SIP -> WebRTC). Also don’t forget to manage the routing in this case.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The most important configuration for a SIP server is its network address (IP and/or domain name). Also you must specify the port number if your server is not using the default SIP port (UDP 5060). This can be done by just appending it after the IP after a colon (such as 11.22.33.44:5678) or as the “port” field in the user table.</span><i><span style='font-size:12.0pt;line-height:115%'> </span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>Note: if you enter the :port after the IP then the port part will disappear, but the port field will be set accordingly.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You <b>don’t need to change any setting on your existing SIP server</b>/PBX/Softswitch as you will see all the traffic coming from the WebRTC clients as normal SIP traffic, usually with the SIP standard digest authentication. The only notable difference is that all traffic from WebRTC clients will come from the Gateway IP, so make sure to white-list this IP on your server or release any rate-limiter restrictions, but do not set IP based authentication for the Gateway IP! (The registrations and calls from WebRTC clients will do proper SIP digest authentication, so you will need to authenticate them as endusers/extensions and not as a SIP trunk with IP authentication, unless you do this intentionally, to unconditionally accept any traffic from WebRTC).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>More details about configuration can be found </span><a href="start.htm#_Add_SIP_Server"><span style='font-size:14.0pt; line-height:115%'>here</span></a><span style='font-size:14.0pt;line-height: 115%'> and </span><a href="start.htm#_Outbound_Routing"><span style='font-size:14.0pt; line-height:115%'>here</span></a><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%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373671"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>WebRTC Client Configuration</span></a></h3> <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%'>In short: just <b>use the “Client configuration” from the “Help” menu</b> for the exact details.</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%'>Details:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Once you finished with the configuration wizard and your gateway is running, you can connect with your favorite WebRTC client and make calls.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Parameters:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>“Gateway address” means your MRTC gateway IP or domain name (if you have assigned a domain)</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>“Mainport” means the gateway websocket port (use the secure port if you enabled SSL)</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>If you have SSL enabled, use WSS for websocket. Otherwise use WS.</span></i></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>SIP domain/proxy/realm: your SIP server domain name or IP (for example 11.22.33.44:5060)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>WebSocket Server URL: the gateway wss://address:mainport/mfstwebsock (for example: wss://domain.com:443/mfstwebsock)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>TURN server address: the gateway address and the main unsecure port (for example domain.com:80)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>STUN server address: the gateway address and the <span style='color:#984806'>mainaportudp</span> port (for example domain.com:8090)</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%'>To find out how to configure your WebRTC client go to the “Help” menu and select the “Client configuration” item. Use the details listed in the “#### WebRTC” section. This will list all you need to know to connect with various WebRTC clients.</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'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'>Note: if you haven’t set a domain and SSL certificate for the gateway or your webpage is not on HTTPS (secure HTTP with valid SSL certificate) then WebRTC will not work from Chrome. In this case you might use Firefox for tests.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'> </span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For the first test call use two enduser accounts (extensions) valid on your SIP server and make calls between them. Use the first account in your WebRTC client (such as webphone, sipml5 or jssip) and the second account in a SIP client (such as MizuPhone or X-Lite). Register with both and call from WebRTC to SIP then from SIP to WebRTC. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Please note that you can also make calls between WebRTC and WebRTC and also between SIP to SIP endpoints (although this latter usually doesn’t make much sense, because there are no reasons why you might connect with SIP clients via the gateway, since your SIP server can handle these anyway).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>During a call, you can open the “Current calls” form in the MizuManage to see the details.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>After the call you can see the CDR (call detail report) by opening the “CDR” form in the MizuManage. If there are no CDRs (call detail records), it means that the call has not reached the server (wrong network settings on server or client side).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373672"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Troubleshooting</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>If you can’t connect/register or make calls, check the followings:</span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>Make sure that the service is actually started (check the “mserver” service status in Windows Service manager)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Make sure that the application is enabled on your Firewall. It should enable itself automatically, however if you are using some third-party firewall or it doesn’t have enough rights, then you might have to enable manually the following executables: mserver.exe, lego.exe, tlsproxy.exe and \tsmodules\fs\fs_mserver.exe</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Have a look at the “Dashboard” form</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Open the “Analyze” form and click on the “Analyze” button to detect potential issues</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Check any important errors or warnings on the “Logs” form</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Make sure that you have basic connectivity to the important ports such as 80 and 443. Use telnet gatewayip port command from a PC from where you are trying to connect with WebRTC</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Install a SIP softphone (such as </span><a href="http://www.counterpath.com/x-lite-download/"><span style='font-size:14.0pt'>X-Lite</span></a><span style='font-size:14.0pt'> or </span><a href="https://www.mizu-voip.com/Portals/0/Files/MizuPhone_Install.exe"><span style='font-size:14.0pt'>MizuPhone</span></a><span style='font-size:14.0pt'>) on the same server where you are running your Gateway and make sure that you can make call to your SIP server from there.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>If you are making call to a user/extension, make sure that the called user is registered when you call it</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>If you don’t hear any voice you might change the RTP routing for the user(s) to “always route RTP” from MizuManage -> Users and devices -> Edit tab</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>If the call setup time is too long, check </span><a href="start.htm#_Slow_call_setup"><span style='font-size:14.0pt'>here</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><span style='font-size:14.0pt'>Are you running your WebRTC or SIP endpoint on the gateway PC? Don’t do this. Most WebRTC stacks has various bugs when they are run from the same IP where they connect to.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>In case of call failure you can check the disconnect reason from the “CDR” form</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Verify the log files. You can find the logs in the server app folder (near the mserver.exe) -> “logs” subfolder. You can also open the folder from MManage -> View Menu -> Files -> Server logs.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Open the last log file (“log_xxx.dat”) with a fast text viewer (For example F3 from Total Commander). To find application errors, open the last log file in the server app directory and search for “ERROR” and “WARNING”. To find a call, search for “INVITE sip:callednumber”.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>You can modify the trace details with the “loglevel” configuration option: from 1 to 5.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Contact </span><a href="mailto:info@mizu-voip.com?subject=WebRTC-SIP%20Gateway%20support"><span style='font-size:14.0pt'>Mizutech support</span></a><span style='font-size: 14.0pt'> if the problem persists and you have or wish to buy a license</span></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> <h1 style='margin-top:0in'><a name="_Toc478373673"><span style='font-size:16.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Management and </span></a><a name="_Toc285322907"><span style='font-size:16.0pt;line-height:115%;font-family: "Calibri",sans-serif'>Monitoring</span></a></h1> <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 most important settings can be configured from Tools menu -> Configure -> Configuration Wizard as already discussed in the previous chapter.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Once the WebRTC-SIP gateway is installed and configured correctly, there is nothing much to manage on the Gateway itself as it has full auto-management capability (such as auto deleting log files, auto fine-tune to your hardware, automatic user management and others).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>This means that this whole chapter is completely optional and you might not need to know about these details at all to run a fully working WebRTC-SIP proxy.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373674"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>MizuManage</span></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'> </span></h3> <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%'>All administration and monitoring tasks can be done from the <b>MizuManage</b> (MManage / MizuManagement) admin client, which is included in the WebRTC gateway install package.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Using the Gateway, you will still need to use your Softswitch/PBX to manage your Users (authentication, authorization, and accounting), Routing, Billing and any other aspects of your network.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Regardless of this, the WebRTC-SIP proxy also provides you complete monitoring and GUI based configuration managements, so you can easily change settings or find connectivity issues if that is WebRTC-SIP conversion related.</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%;color:#984806'>Always check the status bar (the bottom text display). MManage rarely displays popups and the success/error statuses of various operations are displayed on the status bar instead.</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%'>Filters:</span></u></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Nearly all forms in MManage can be filtered with the followings:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Quick filter: found in the top-left side in MizuManage. For example type “44*” in the quick filter box then open the “CDR” form and click “Load”. You should be able to see all calls to 44….. numbers. Or enter “test” and open the “Users and devices” form. Click on the load button to see accounts containing the “test” word (in name, username, address, etc)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Direction filtering: accessible by double clicking on the space above the quick filter or from the View menu -> Filter -> Set direction filter. When you are doing operation which needs more precision (eg. billing), always use the Set Directions form and not the quick filter.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Date-Time filter: found in the top-left side in the MizuManage. Useful to restrict statistics, reports and CDR listing intervals.</span></p> <p class="MsoListParagraphCxSpLast" 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="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><u><span style='font-size:14.0pt;line-height:115%'>Menu:</span></u></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>The most important item is the Configuration wizard (from Tools menu -> Configure). Use this to (re)configure your gateway most important 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'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>From the “Control” menu you can Stop/Start/Restart your server (you can do so also from windows Services management console)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>To export data from the application, use File menu -> “Save As”. </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>From the “Fields” menu you can manipulate the selected dataset (grids, etc).</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>The Tools menu -> “MManage Settings” form is used for local MManage admin client configuration only.</span></p> <p class="MsoListParagraphCxSpLast" 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> <h3 style='margin-top:0in'><a name="_Toc478373675"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>User Management</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>All user records can be managed from the “Users and devices” form (below the “Access” node in the tree-view).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Users can represent real people, endpoints, extensions, devices or servers.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The most important user types are the followings:</span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>SIP server for outbound routing. Your SIP server(s) should be added here.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Traffic senders for inbound routing. Your SIP server(s) should be added also here.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Endusers: these are managed by the Gateway automatically.</span></p> <p class="MsoListParagraphCxSpLast" 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><span style='font-size:14.0pt'>Admin users: you should add yourself (and/or the administrator) here as it is used for various purposes such as sending alerts and reports.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The users are stored in the database tb_users table and they have a lot’s of properties which can be managed from the “Users and devices” form or (for advanced users only) changing database record fields (select a user the go to Fields menu -> Show fields).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><b><span style='font-size:14.0pt;line-height:115%'>You don’t need to manage endusers/extensions on the Gateway!</span></b><span style='font-size:14.0pt; line-height:115%'> In MManage you might see your endusers (extensions) listed on the “Users and devices” form, however that is only for monitoring purpose and for the gateway internals, auto created and managed by the Gateway. Except your extension usernames, no authentication credentials (password) or any sensitive information (such as billing) are stored on the Gateway. Use your Softswitch/PBX for user management.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><b><u><span style='font-size:14.0pt;line-height:115%'>Listing users</span></u></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Open the “<b>Users and Devices</b>” form (below the “Access” section) and (double) click on a user type to have a listing. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can apply various filtering using the user “Type” checkbox-list, the dropdown-list on the top of the form or the already discussed direction filter or quick filter.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Your SIP servers can be found as “SIP Server” and “Traffic sender” accounts.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'>You can easily search for users using the “quick filter” box. For example to list all outbound routes whose username or name contains the “carrier” word, select the “SIP Server” and type “carrier” in the quick filter. The quick filter will also search in other fields such as IP, name, email and others.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'>Other listing options are available also from right click on the “Users and devices” node in the main tree-view or right click on user type node from within the “Users and devices” form.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><a name="_Toc285322900"><b><u><span style='font-size:14.0pt;line-height:115%'>Creating users</span></u></b></a></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The best way to create new users is to clone an existing working account with the same user type.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For this, launch the “<b>Users and Devices</b>” form, select a user type, and click on the “Load” button. Then select any user entry and click on the “New User” button. <span style='color:black'>Alternatively right click on the “New” button for other options to add user records</span><span style='color:#984806'>. </span></span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Endusers</span></b><span style='font-size:14.0pt'> are created automatically by the server at first time when an extension will try to register or call. These enduser records are only for internal management and statistics reasons. This are the same users as the extensions on your PBX, but the Gateway might have information only about their username and display number (the gateway does not have to know the password since the authentications are forwarded to your SIP server). You don’t need to manage these endusers on the gateway (Keep using your PBX to manage your users).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The username can be also used as a real phone number; will be used also as the caller-id if not specified otherwise. Endusers can make voice or video calls, presence and chat directly between them without the need to be forwarded to your SIP server. (Alternatively you can disable direct routing between endusers by setting the <span style='color:#984806'>alloweuserusercalls</span> and <span style='color:#984806'>fs_directroutrfstofs</span> global config values to 0).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Traffic sender</span></b><span style='font-size:14.0pt'> users are used for receiving traffic from your SIP servers (PBX, Softswitch, SIP proxy or other SIP device). The authorization type is usually set to “Auth ip must match” and you have to enter a correct “Auth Ip” (IP address based authentication). </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>SIP servers</span></b><span style='font-size:14.0pt'>: For outbound traffic you need one (or more) SIP Server user. The most important parameter here is the “IP” where the VoIP calls will be sent. Then you will have to configure these also on the Routing form. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>To be able to send and receive traffic to/from another SIP server or carrier you will have to add it as both a “traffic sender” and “sip server” user.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373676"></a><a name="_Add_SIP_Server"></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Add SIP Server</span></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Your upper SIP server(s) can be added multiple ways:</span></p> <p class="MsoListParagraphCxSpFirst" 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'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>from the Configuration wizard you can add your default/first SIP server on the “SIP server” page</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 style='font-size:14.0pt'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>from the “Add Upper SIP Server” form (from Tools menu -> Add Import users)</span></p> <p class="MsoListParagraphCxSpLast" 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'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>from the “Users and devices” form (This is discussed here)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>A default/first SIP server can be added during the configuration wizard. (You can also modify it from right click on “Configurations” node in the tree-view and select the “Upper server” from the popup menu).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The WebRTC-SIP proxy can handle multiple upper servers in case you have more SIP servers where you wish to accept WebRTC traffic.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>To add a SIP server open the “Add Upper SIP server” from Tools menu -> Add Import users.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>(Alternatively right click on the “Users and devices” node in the tree-view and select “Add Upper SIP Server”).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Pay attention to the following fields: IP, domain, port (you can also set the proxy field if needed).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Once you added your server, make sure to </span><a href="start.htm#_Outbound_Routing"><span style='font-size:14.0pt'>modify the routing</span></a><span style='font-size:14.0pt'> after your needs (if you have more than one server, then the routing needs to be able to decide somehow which traffic to send to which server. You can create various rules for this on the “Routing” form)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>When you add a SIP server, this is what happens:</span></p> <p class="MsoListParagraphCxSpFirst" 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'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>a Traffic sender account is created which will handle inbound traffic coming from your SIP server (SIP -> WebRTC)</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 style='font-size:14.0pt'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>a SIP server account is created which will handle outbound traffic to your SIP server (WebRTC -> SIP)</span></p> <p class="MsoListParagraphCxSpLast" 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'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>a default routing entry is created for the above SIP server</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Actually, you can “add a SIP server” also by just adding the above accounts manually from “Users and devices”.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>If the WebRTC clients have to register to different SIP servers, set the “<span style='color:#984806'>routingforregister</span>” global config option to 1 (from the “Configurations” form). Otherwise set to 0. If the “<span style='color:#984806'>routingforregister</span>” is set to 0, then it will forward all registrations to your default SIP server which you have created during the configuration wizard.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>More details about multiple SIP servers can be found </span><a href="start.htm#_Multiple_SIP_servers"><span style='font-size: 14.0pt'>here</span></a><span style='font-size:14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373677"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Add User</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>There is <b>no need</b> to add endusers, extensions or WebRTC clients manually. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Just continue to manage your users as before: on your SIP server.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You will be able to see the users on the “Users and devices” form -> Endusers, however these are managed automatically by the Gateway and there is no need to change any settings here except if you have some extra requirement such as setting call forwarding, call recording or some other per user setting. The first time a WebRTC client will register or make call, a user entry is automatically created by the Gateway. This is mostly for internal usage only, but it also allows you to make per user changes if necessary. No password of these users are stored on the WebRTC gateway (The users are created with “Username” based authentication only, however the gateway will also check other parameters such as same session / same address for further account restriction. If you wish, you can create additional users of your own for various other purposes.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For special purposes you might add users also manually. There are multiple ways to do this: </span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>click on the “New” button from “Users and Devices” </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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>or right click on the “New” button </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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>or right click on the “Users and devices” tree-view node.</span></p> <p class="MsoListParagraphCxSpLast" 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><span style='font-size:14.0pt'>or from “Tools” menu -> “Add or import users”</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:#984806'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373678"></a><a name="_Toc285322902"></a><a name="_Outbound_Routing"></a><span style='font-size:14.0pt;line-height:115%; font-family:"Calibri",sans-serif'>Outbound Routing</span></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Outbound routing rules are required for the gateway to properly route the calls to your SIP server(s) (for WebRTC -> SIP calls).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Calls between users are handled automatically as this will bypass the routing table and it is routed directly </span><a href="start.htm#_Calls_between_endusers/extensions"><span style='font-size:14.0pt'>unless if you disable it</span></a><span style='font-size:14.0pt'>. However for outbound routing you need to add a “SIP server” user first and then add it to your routing rules.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Once you add your upper server as a “SIP server”, open the “<b>Routing</b>” <b>form</b>. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>In the left side you have to define your pattern which will restrict the condition when the actual route entries can be used. If all fields are empty and the time definition is set to “All times” then all patterns will match. You can make restrictions if you make specifications here (caller, called prefix, time restriction, etc). Make sure that you increase the priority for the pattern (to be higher than the “general” pattern where you have not made any restrictions).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>On the right side you will have to add one or more sip proxy users. If you set more than one route with equal priority, then you have load balancing, LCR or BRS (depending on the “<span style='color:#984806'>brs_lcr</span>” global config option); otherwise the traffic will be routed after the prioritizations (will flow to the lower priority servers only if you have reached the maximum port limitations or because automatic failover). </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:#984806'>If you have only one server then just create one pattern (left side) enabled for all times and no any filter set, then assign your SIP server user entry to this pattern (right side).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:#984806'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For more details please read the </span><a href="https://www.mizu-voip.com/Portals/0/Files/Routing_guide.pdf"><span style='font-size:14.0pt'>routing guide</span></a><span style='font-size:14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><b><u><span style='font-size:14.0pt;line-height:115%'>Dial plans</span></u></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can manipulate number format, SIP headers or the Caller-ID from the following settings:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-bottom:0in;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><span style='font-size:14.0pt'>Users and devices form: caller-id, username, other numbers (DID match), tech prefix</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;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><span style='font-size:14.0pt'>Routing form: you can only specify routing direction here without number changes</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;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><span style='font-size:14.0pt'>Rules form: this is a powerful module which you can use to change almost anything (including caller-id, called number format and many others)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;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><span style='font-size:14.0pt'>Global configuration: a few global configuration options might also affect the dial plan. Right click on the “Configuration” node and select “Number rewrite” for the details.</span></p> <p class="MsoListParagraphCxSpLast" style='margin-bottom:0in;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><span style='font-size:14.0pt'>Prefix rules and the dial plan form: use the “Rules” form instead when possible.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can find more details in the </span><a href="https://www.mizu-voip.com/Portals/0/Files/MAdminGuide.pdf"><span style='font-size:14.0pt'>VoIP Admin Guide</span></a><span style='font-size: 14.0pt'> below the Routing section.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373679"></a><a name="_Toc285322904"></a><a name="_Inbound_Routing"></a><span style='font-size:14.0pt;line-height:115%; font-family:"Calibri",sans-serif'>Inbound Routing</span></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>If you would like to accept traffic from your servers (SIP -> WebRTC calls), you need to have one or more “<b>Traffic sender</b>” user account created. Usually you can use IP based authentication. For this, add the peer IP to the “<b>Auth IP</b>” field.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For each incoming call, the server will first check if the called party is a local user. If not, then the call is routed according to the rules which are set by the “Routing” form.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'>One “Traffic sender” user entry can handle incoming traffic with IP authentication from multiple IP addresses. Use the “…” button near the “Auth IP List” to add more IP addresses.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'>Usually there is no need to setup any routing rule for inbound calls (as these will be calls to local WebRTC users which are found automatically).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'>Traffic between endusers/extensions is handled automatically.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373680"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>CDR</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>A CDR (call details report) is generated for each call on your gateway.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can access these records by using the <b>“CDR</b>” form under the “Monitoring” node in the tree-view. By default only the most important fields are listed (date-time, connect time, call duration, etc). You can see more details if you check the “All fields” checkbox. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>To quickly list the CDRs that belong to a user, open the “Users and devices” form. Find the user record, then right-click on it and select “Set Direction”. Than go back to the CDR form and click on the “Load/Reload” button.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'>You can easily filter calls with a specific prefix by typing the prefix in the quick search box following with an asterisk and hit enter. For example searching for 44* will list all CDR where the called number begins with 44.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'>You can also see various statistics based on these CDR by using the “Advanced Statistics” or the “Disc. Reasons” forms.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:black'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373681"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Statistics and Monitoring</span></a></h3> <p class="MsoNoSpacing"><span style='font-size:14.0pt'> </span></p> <p class="MsoNoSpacing"><span style='font-size:14.0pt'>The Mizu WebRTC-SIP proxy provides endless possibilities for monitoring both real-time and statistics history. Some of the most important tools are the followings:</span></p> <p class="MsoNoSpacing"><b><span style='font-size:14.0pt'>Dashboard</span></b><span style='font-size:14.0pt'>: a summary of the most important parameters and a start point for management (Note: You can access various statistics by just clicking on the Dashboard items. For example click to “CCalls” will show the current calls)</span></p> <p class="MsoNoSpacing"><span style='font-size:14.0pt'>By using the “<b>Analyze</b>” form you can have a quick overview about the system and warnings for malfunctions or configuration suggestions</span></p> <p class="MsoNoSpacing"><b><span style='font-size:14.0pt'>List the active sessions: </span></b><span style='font-size:14.0pt'>Monitoring -> “Current Calls” form</span></p> <p class="MsoNoSpacing"><b><span style='font-size:14.0pt'>Disconnect reasons</span></b><span style='font-size:14.0pt'>: Monitoring -> Disc. Reasons will show statistics about call disconnect codes.</span></p> <p class="MsoNoSpacing"><b><span style='font-size:14.0pt'>Statistics by SIP server: </span></b><span style='font-size:14.0pt'>Monitoring ->Advanced Statistics -> Group By: called servers (useful if you have multiple server, softswitch or PBX)</span></p> <p class="MsoNoSpacing"><b><span style='font-size:14.0pt'>Statistics by users: </span></b><span style='font-size:14.0pt'>Monitoring ->Advanced Statistics -> Group By: caller</span></p> <p class="MsoNoSpacing"><b><span style='font-size:14.0pt'>Statistics by day: </span></b><span style='font-size:14.0pt'>Monitoring ->Advanced Statistics -> Group By: day</span></p> <p class="MsoNoSpacing"><b><span style='font-size:14.0pt'>Logs</span></b><span style='font-size:14.0pt'>: the most important logs can be listed from the “Logs” form. Detailed logs can be found in log files accessible from View menu -> Files -> Server Logs directory</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Other more <b>advanced statistics</b> can be generated by using the Advanced Statistics form and using different fields/options/grouping/directions.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>All statistics can be filtered by the “set direction” form or the “quick filer” edit-box and by a time interval selection.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Automatic reports: </span></b><span style='font-size:14.0pt'>The MRTC Gateway can send daily reports for administrators or email/sms alerts on malfunctions. For this you have to setup an “Admin” user with a valid email address (From the “Config Wizard” or from “Users and Devices”). Then set the following user fields to 1 (after your needs): “<span style='color:#984806'>sendemailalert</span>”, “<span style='color:#984806'>senddailyemal</span>”, “<span style='color:#984806'>sendmonthlyemail</span>”, “<span style='color:#984806'>sendsmsreport</span>”, “<span style='color:#984806'>sendsmsalert</span>”.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373682"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Start/Stop</span></a></h3> <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%'>You can start/stop the WebRTC-SIP gateway service from:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>the “Control” menu in MManage </span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>or by the Windows Service manager locate the “mserver” entry, right click and select “Start” or ”Stop” </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'><i><span style='font-size:12.0pt;line-height:115%'>Note: the first start might take a bit longer (around one minute), otherwise the start-up time should be below 10 seconds.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%'> </span></i></p> <h3 style='margin-top:0in'><a name="_Toc478373683"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Backup/Restore</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>A backup for MRTC can be done with simple file copy (xcopy) and just copy back the files for restore.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can back-up all your data and settings by just copying the mserver.sdf file from the app folder</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Alternatively, you might choose to copy the entire app directory, excluding logs (however these files can be easily recreated by a reinstall, so you really need only the mserver.sdf).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>By default the Gateway will automatically create nightly backups and auto-delete old backups (Search for “backup” in the global configuration to change this).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h1 style='margin-top:0in'><a name="_Toc478373684"></a><a name="_Additional_Settings"></a><span style='font-size:16.0pt;line-height:115%; font-family:"Calibri",sans-serif'>Additional Settings</span><span style='font-size:16.0pt;line-height:115%;font-family:"Calibri",sans-serif'> </span></h1> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The below discussed settings have to be changed only if you have some special requirements, otherwise the default settings are optimal values for a successful WebRTC-SIP protocol conversion.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Through this guide we often refer to various configuration keys also called as “global configuration” or “global config”. All these can be managed from the “Configuration” form (below the “Other” node in the tree-view).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373685"></a><a name="_Toc478281594"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Settings</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The gateway has a long list of configurable settings which you might adjust to turn on/off features or to adjust the WebRTC-SIP proxy behavior after your needs.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The settings can be categorized in the following way:</span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>global configuration (editable from the “Configuration Wizard” and from the “Configuration” form)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>per user configuration (editable from “Users and devices”)</span></p> <p class="MsoListParagraphCxSpLast" 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><span style='font-size:14.0pt'>other configurations (editable form various forms such as “Routing”, “Rules” and others)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373686"></a><a name="_Toc478281595"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Global Configuration</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can quickly change any global configuration from the “Configuration” form (below the “Other” node from the main tree-view). Just search for your keyword to find the related setting(s).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Most of the changes can be applied instantly (click on the “Apply Now” button), however some changes (such as local/bind ip/port reconfiguration) require a restart.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Some settings groups can be accessed by right clicking on the “Configuration” form. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>SBC related options can be listed by right click on the “Configuration” node and select the “Gateway” from the popup. Some important gateway related settings are the following:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>autocreatereguse:</span><span style='font-size:14.0pt;line-height:115%'> 0=no,1=when fwd authenticated ok register, 2=always (when we receive the register)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>fwdregistration:</span><span style='font-size:14.0pt;line-height:115%'> 0=no,1=only from alternate port, 2=always</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>fwdregistrations_address</span><span style='font-size:14.0pt;line-height:115%'>/<span style='color:#984806'>fwdregistrations_domain</span>/<span style='color:#984806'>fwdregistrations_ip</span>/<span style='color:#984806'>fwdregistrations_port</span>: the address of your SIP server (the upper server)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>forwardauthentifications:</span><span style='font-size:14.0pt;line-height:115%'> 0=no (default),1=yes,2=yes with username as callername, 3=yes with phonenumber as callername, 4=yes with username as authname too, 5=yes with phonenumber as authname too, 6=replace authorization with username but leave the A number intact, 7=replace authorization with sipphonenumber</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>forwardauthpassword</span><span style='font-size:14.0pt;line-height:115%'>: 0=fwd from ep, 1=means 2 if from webrtc for recent users, 2=answer from local always,3=answer from local also for register, 4=answer from remapped upperusername/upperpassword</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>fs_restrictsiptolocal</span><span style='font-size:14.0pt;line-height:115%'>: calls from SIP will be allowed only to webrtc. 0=no,1=yes if mizu gateway (def),2=yes if gw, 3=yes always (2 and 3 should not be used)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>fs_checkdomainmismatch</span><span style='font-size:14.0pt;line-height:115%'>: useful if you have multiple servers with different users and wish to avoid mismatch routing. 0=no,1=yes (def if fs_ismzgateway > 1),2=always</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373687"></a><a name="_Toc478281596"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Clean Install</span></a></h3> <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%'>By default when you reinstall the MRTC service, it will keep your old settings and data, thus you can easily upgrade to new versions.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If you wish to begin with a clean state, delete (backup!) the old <span style='color:#984806'>mserver.sdf</span> file before launching the installer, thus you will have a clean state with all your previous settings and data erased.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373688"></a><a name="_Toc478281597"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Domain name</span></a></h3> <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%'>Certain browsers require HTTPS for WebRTC to work (both for your website and for Websocket/DTLS) and to be able to setup a valid SSL certificate for your WebRTC-SIP proxy, first you need to assign a domain name to its IP address.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The domain name is also useful if later you wish to change your Gateway IP address or location (in this case you will just need to redirect the domain to the new IP and everything will continue to work as previously).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If you don’t already have a domain, you can purchase it easily from any domain registrar company such as </span><a href="https://godaddy.com/"><span style='font-size:14.0pt;line-height:115%'>GoDaddy</span></a><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%'>If you already have a domain (for example created for your website), then you can easily add a sub-domain as most domain registrars provides an online user interface where you can add sub-domains for free.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>A sub-domain is perfectly fine for the WebRTC-SIP gateway, for example rtc.yourdomain.com. The sub-domain must have an A record set to your Gateway IP (ipv4 address).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Also the WebRTC gateway can provide you a free domain name with a single mouse click: on the Configuration Wizard click on the “get” link near the domain name to obtain it (provided by Mizutech auto-dns service). Please note that it might take some time until the domain will be assigned for your server (sometimes instantly, but sometimes it might take 2 or more hours). However, in production it is a good idea to assign your own domain name (which is owned and controlled by you). </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can use the <span style='color:#984806'>nslookup</span> or the <span style='color:#984806'>ping</span> tools from your command line to see if the domain assignment is correct.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:12.0pt;line-height:115%;color:#984806'>For example:</span></p> <p class="MsoNormal" style='margin-top:0in;margin-right:0in;margin-bottom:0in; margin-left:.5in;margin-bottom:.0001pt'><span style='font-size:12.0pt; line-height:115%;color:#984806'>nslookup rtc.yourdomain.com</span></p> <p class="MsoNormal" style='margin-top:0in;margin-right:0in;margin-bottom:0in; margin-left:.5in;margin-bottom:.0001pt'><span style='font-size:12.0pt; line-height:115%;color:#984806'>ping rtp.yourdomain.com</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>See if the response lists your MRTC Gateway IP. If you have recently assigned the domain, then you might have to wait a few hours for the dns cache propagation. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Your OS might also cache (incorrect/old) dns record. Use the <span style='color:#984806'>ipconfig /flushdns</span> to clear the cache and recheck again with one of the above commands (<span style='color:#984806'>ping</span> or <span style='color:#984806'>nslookup</span>).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373689"></a><a name="_Toc478281598"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>SSL certificate</span></a></h3> <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%'>If you are using a web based WebRTC client (not a native library) then most probably you will need an SSL certificate. This is because certain browsers such as Chrome and Opera will allow WebRTC only if you are using secure websocket (WSS). Self-signed SSL certificates will silently fail, so you will need a valid certificate. Also your webpage from where you launch your WebRTC client need to be hosted on HTTPS (or test from local file).</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'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>“SSL certificate” is a commonly used name but in fact it really means signed RSA keypairs used for TLS encryption.</span></i></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 WebRTC gateway can acquire and maintain a <b>free Let’s Encrypt certificate automatically</b> which is just perfect for WebRTC.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>To enable this, make sure to select the “Auto get SSL/TLS certificate” checkbox on the “Network” page of the Configuration Wizard (or set the “<span style='color:#984806'>autotlscert</span>” global config to 1).</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%'>The following are the prerequisites for the auto-ssl to work:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>the gateway must have internet access </span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>you must have a valid domain name assigned to your gateway IP (discussed in the previous point)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>ports 80 and 443 must be available for the Gateway (not used by other apps such as a Web server)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>4.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>if the gateway is behind NAT, make sure to forward ports 80 and 443 correctly</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'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'>Note: starting from MRTC v.2.4 (2018 Apr.) there is no need to forward ports 80 and 443 anymore if you acquired a webvoipphone.com subdomain since the new version is capable to utilize already prepared wildcard certificate for this domain.</span></i></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> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If you prefer, you can also use <b>your own SSL certificate </b>if you already have one and it is assigned for the (sub)domain which you have set for the Gateway. For this you just have to copy the following files in the gateway directory:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>key.pem</span><span style='font-size:14.0pt;line-height:115%'>: the key file (your private key which you have got when generated the CSR - keyfile.key)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>cert.pem</span><span style='font-size:14.0pt;line-height:115%'>: the certificate file (your certificate as received from the CA -yourdomaincertificate)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>root.pem:</span><span style='font-size:14.0pt;line-height:115%'> the chain: CA intermediary + root certificate. intermediate.crt + root.crt</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%'>You can use the Gateway also <b>without an SSL</b> certificate. In this case use unsecure websocket (ws:\\IP:port) from any native WebRTC client or Firefox (It will not work from Chrome and Opera).</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%'>SSL certificate on local LAN:</span></u></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Since it is impossible to acquire a valid certificate for a local domain assigned to a private IP address, you can do the following workaround in case if you wish to run MRTC on a private network:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>1. Create a (sub)domain and assign it to any existing web server IP and request a certificate for it (since you can easily request TLS certificates only for public webservers)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>2. Copy the certificate files to the MRTC app directory (.pem files as described above)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>3. Redirect the above (sub)domain to your gateway private IP address (just rewrite it’s A record settings)</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'><i><span style='font-size:14.0pt;line-height:115%;color:#984806'>Note: starting from MRTC v.2.4 (2018 Apr.) you can just get a webvoipphone.com subdomain for your private IP since the new version is capable to utilize already prepared wildcard certificate for this domain.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373690"></a><a name="_Toc478281599"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Change IP</span></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'> address</span></h3> <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%'>If you migrate the WebRTC-SIP Gateway to another box or your IP changed, then you will need to set the new IP.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>There are 2 ways to easily change the Gateway IP:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>going through the Configuration Wizard and make changes on the “Network” page</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>or right click on the “Configuration” node in the main tree-view (below the “Other” node), select “Network -Basic” and rewrite the IP</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Restart your gateway once you changed the IP because this can’t be applied at runtime.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373691"></a><a name="_Toc478281600"></a><a name="_Ports"></a><span style='font-size:14.0pt;line-height:115%;font-family: "Calibri",sans-serif'>Ports</span></h3> <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%'>Since VoIP servers are using a wide range of ports, it is recommended to host it on a public server (with a public IP), unless you plan to use it only from an internal network. In addition, it is recommended to turn off any external or third party port based firewall (you can still use the Windows built-in firewall where it is easy to set exceptions for applications and not for ports).</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%'>This section is important if you are hosting the WebRTC-SIP proxy behind a NAT or firewall for some reason and the gateway is used from the external network (such as the public internet).</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%'>The MRTC listen on several ports to offer its services: SIP signaling, Websocket, RTP/SRTP ports range, RTCP/SRTPC port range, TURN, STUN and others.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>It is recommended to keep the default ports since the defaults are optimized for maximum connectivity and/or are conform to standards.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For example it is highly recommended to keep the default TCP port 80 for Websocket, TURN and API and use TCP 443 for the secure version of these. A lot of corporate networks allows traffic only on port 80 and 443 and if your gateway is using other ports the clients from behind more restrictive NAT’s and firewalls will not be able to connect with their WebRTC client. If you are behind NAT, it is recommended to set one-to-one port mapping (internal listening ports mapped to the same external ports on your NAT device).</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%'>If these ports are already used by some software running on the same host, then we recommend to assign a secondary IP for your server and bind the gateway to this new IP (set the bindip), then reconfigure the other software’s to use only the old IP (bind to the old IP). For example IIS can be bound to an IP with the following command: </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>netsh http add iplisten ipaddress=IPADDRESS</span><span style='font-size:14.0pt;line-height:115%'> (replace IPADDRESS with the IP to bind)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If you are hosting the WebRTC-SIP Gateway on the same box where your softswitch is running, then assign a separate IP (bindip) to the Gateway if possible. If this is not possible, then make sure to change the ports on the gateway to avoid conflicts (if your SIP server is listening on 5060 on the same host, then set a different “<span style='color:#984806'>localport</span>” for the Gateway).</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%'>Here is the <b>list of ports</b> used by the gateway and their global config options which you can change from the “Configuration” form if needed.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Mandatory ports:</span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>localport UDP and TCP (main SIP signaling port. Default is 5060)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>mainaport UDP and TCP (main server port for various purposes websocket, TURN, API and others. Default is 80.)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>mainaportudp UDP (used for STUN, UDP RTP relay and others. default to 8090)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>sslportmain and cfg_sslport TCP (443 by default for SSL, HTTPS, WSS)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>MinRTP-MaxRTP range UDP (for RTP and RTCP; you might enable also TCP for TCP ICE candidates)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>fs_minrtp- fs_maxrtp range UDP (SRTP and SRTCP for WebRTC and RTP for extra PBX)</span></p> <p class="MsoListParagraphCxSpLast" 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><span style='font-size:14.0pt'>tcpcandidatesrvport (TCP relay port, 10080 by default)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Optional ports (good to allow also these):</span></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>Remote desktop TCP 3389 or 3386 (for easy server administration)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>TCP 1433 and/or 2223 (for remote management)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>adminport TCP (for remote CLI access from MManage server console. Default is 9885)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>monitorport TCP (to easy access logs from remote MManage. Default is 9889)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>forwarderport (TCP forwarder for various internal services, 11080 by default)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>localport+1 TCP (for secure SIP sips. Default is 5061 -if you enable TLS for SIP which is known as SIPS)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>ftpserverport TCP (for remote file access. Default so 9710)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>UDP: 44444 (“voice-here” functionality in MManage)</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>TCP: 9886, 9889 (optional ports for admin console and logs)</span></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;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Make sure to enable these ports if you are using a port based firewall.<br> Make sure that the ports used by the gateway are not used also by some other application such as a local 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="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>In short, you should enable/forward the following ports on your firewall/NAT:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>UDP: 80, 5060, 8080, 8090, rtp port range defined by MinRTP-MaxRTP and fs_minrtp- fs_maxrtp</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>TCP: 80, 443, 2223, 3386, 5060, 5061, 9710, 9885, 10080, 11080</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The port forwarding can be quickly tested from MManage -> Tools menu -> Configure -> Tests -> Check port access.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373692"></a><a name="_Toc478281601"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Gateway behind NAT</span></a></h3> <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 WebRTC-SIP gateway can be used also behind NAT (located behind NAT or router, even without internet access).</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'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>Note: this section is not about WebRTC clients connecting from behind NAT. That is a normal use case and the gateway will handle those very well by default, without any special attention needed. This chapter is when you install the Gateway on a local LAN with SIP and WebRTC clients on the same NAT or from the internet.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'> </span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If your gateway is behind a NAT or Router, make sure to forward the </span><a href="start.htm#_Ports"><span style='font-size:14.0pt;line-height:115%'>ports</span></a><span style='font-size:14.0pt;line-height:115%'> above correctly if you need connectivity also from external network <i>(WebRTC clients or SIP server on the internet. If both of these are behind NAT, then there is no need to enable these ports except ports 80 and 443 if you need auto-generated SSL certificate).</i> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>In this case make sure to set the “<b>Offer services for</b>” option correctly on the “Network” page of the Configuration Wizard. </span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Both LAN and internet: select this if WebRTC clients might connect also from the public internet (make sure to set proper port forward on your router in this case)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>LAN: means all WebRTC clients will connect from local LAN, but the SIP traffic might be sent to the public internet (your SIP server is outside)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Force LAN Only: means that all peers (including your SIP server and all clients) are located on the local LAN</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%'>If your gateway is behind NAT and no access to the internet or you don’t forward ports 80 and 443 to it, then you will not be able to use its auto-ssl capability and no valid SSL certificate will be assigned. In this case you will not be able to use WebRTC from Chrome, however Firefox will work just fine over unencrypted websocket. (use ws:\\gatewayIP). This is except if you use a webvoiphone.com subdomain since for this domain there is already prepared wildcard certificate for your convenience.</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%'>In other words, here are the <b>possible uses-cases</b>:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><u><span style='font-size:14.0pt;line-height:115%'>Gateway on the public internet:</span></u></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%'>You can just skip this chapter as you don’t need to take care about NAT in this case. (WebRTC clients connecting from behind NAT are handled automatically)</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%'>Set the “Offer services” option to “Internet only” if your SIP server is also located on the public internet or “Both LAN and Internet” of your SIP server or other SIP device is located on the same box or network.</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%'>Your “Public IP” and “Bind IP” can be the same public address in this case.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><u><span style='font-size:14.0pt;line-height:115%'>Gateway, SIP server and WebRTC clients all on local LAN: </span></u></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%'>In this case you don’t need to set any port forwarding in this case as all peers will be reachable directly. The only thing to pay attention to is the SSL certificate, since an SSL certificate can be only installed if you have a valid domain assigned to your gateway IP. To be able to use the auto-ssl functionality, you also need to forward 80 and 443 TCP ports except if you use a webvoiphone.com subdomain since for this domain there is already prepared wildcard certificate for your convenience. </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%'>The Gateway can be used also without any certificate via unsecure websocket from Firefox or from native WebRTC clients.</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%'>Set the “Offer services” option to “Force LAN only” in this case.</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%'>Your “Public IP” and “Bind IP” can be the same private address in this case.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><u><span style='font-size:14.0pt;line-height:115%'>Gateway and WebRTC clients behind NAT, SIP server on the internet:</span></u></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%'>If the SIP server or SIP service has good NAT handling capability, then everything should work just fine by default. Otherwise setup proper port forwarding on your NAT/router for the above mentioned ports.</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%'>Set the “Offer services” option to “LAN” in this case.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><u><span style='font-size:14.0pt;line-height:115%'>Gateway behind NAT, WebRTC clients on the public internet:</span></u></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%'>Set the “Offer services” option to “Both LAN and Internet” in this case and setup proper port forwarding on your NAT/router for the above mentioned ports.</span></p> <p class="MsoListParagraphCxSpLast" 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%'>Also set the “Public IP” to your network </span><a href="http://whatismyipaddress.com/"><span style='font-size: 14.0pt;line-height:115%'>external IP</span></a><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%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>Don’t run your WebRTC or SIP client from the same PC where MRTC is running. Most software has various bugs if uses the same local IP as the gateway IP!</span></i></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> <h3 style='margin-top:0in'><a name="_Multiple_SIP_servers"></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Multiple SIP servers</span></h3> <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%'>You can use one MRTC gateway for multiple SIP servers (if you have more servers) for both outbound and inbound 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;line-height: normal'><b><span style='font-size:14.0pt'>Registrar routing:</span></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>By default all registrations will be routed to the default upper SIP server (what you have set during the configuration wizard or from global configuration search for “<span style='color:#984806'>fwdregistrations_</span><span style='color:black'>”</span>).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If you have multiple upper SIP servers and the WebRTC clients need to register to different servers, make sure to set the <span style='color:#984806'>routingforregister</span> global config option to <span style='color:#984806'>1</span> (from the “Configuration” form).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Then the registrations will be routed after the routing rules which can be defined on the “Routing” form. The only exception is the called number since there is no called party in a registration (this is applicable only for calls).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>However, if your SIP servers are for completely different domains/businesses, then you should consider to set-up a separate WebRTC-SIP gateway for each of them.</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'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'>Note: The “routingforregister“ is only for registrations. Call and chat routing are always performed after your “Routing” settings</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:#984806'> </span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Call routing:</span></b></p> <p class="MsoListParagraphCxSpFirst" 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><span style='font-size:14.0pt'>Outbound calls are always routed based on </span><a href="start.htm#_Outbound_Routing"><span style='font-size:14.0pt'>routing rules</span></a><span style='font-size:14.0pt'> which you define on the “Routing” form.</span></p> <p class="MsoListParagraphCxSpMiddle" 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'>Routing rules can be set for SIP servers (so you must create “SIP server” user(s) before to be able to work with routing.</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 style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>To be able to accept </span><a href="start.htm#_Inbound_Routing"><span style='font-size:14.0pt'>inbound calls</span></a><span style='font-size:14.0pt'> (from your SIP server to WebRTC) you will need to create “Traffic Sender” user (usually with IP authentication with the Auth IP set to your SIP server address). </span></p> <p class="MsoListParagraphCxSpLast" 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><span style='font-size:14.0pt'>Calls between users (such as WebRTC to WebRTC) are handled automatically.</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;line-height: normal'><span style='font-size:14.0pt'>More details about SIP servers can be found </span><a href="start.htm#_Add_SIP_Server"><span style='font-size:14.0pt'>here</span></a><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%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373694"></a><a name="_Toc478281603"></a><a name="_Calls_between_endusers/extensions"></a><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Calls between endusers/extensions</span></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Calls between endusers can be routed directly, without the need to route them via your upper SIP server(s), thus saving your server resources (CPU/memory).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>These are usually calls between WebRTC clients (WebRTC to WebRTC), but they can be also WebRTC to SIP or SIP to WebRTC or SIP to SIP calls (if you register also SIP devices on the gateway).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>This direct routing is enabled by default and it can be easily changed by the following global config values (from the “Configurations” form):</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:#984806'>alloweuserusercalls</span><span style='font-size:14.0pt'>: 0=no (don't check if local target),1=yes,2=disable (drop) call to endusers</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt;color:#984806'>fs_directroutrfstofs</span><span style='font-size:14.0pt'>: -1=auto,0=no,1=via mizu server,2=within webrtc stack (route call between webrtc and webrtc within the webrtc stack), 3=quick websocket only,4=auto</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>To enable direct routing, set the <span style='color:#984806'>alloweuserusercalls</span> to 1 and <span style='color:#984806'>fs_directroutrfstofs</span> to 3.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>To disable direct routing, set the <span style='color:#984806'>alloweuserusercalls</span> to 0 and <span style='color:#984806'>fs_directroutrfstofs</span> to 0.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>If you disable direct routing, all calls will be routed via your SIP server(s). If the called endpoint is registered via the MRTC gateway, then the SIP server might route the call back to the gateway, so the call flow might look like this: Caller WebRTC endpoint -> MRTC gateway -> SIP server -> MRTC gateway -> Called WebRTC endpoint</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373695"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Multiple registrations and call fork</span></a></h3> <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%'>One user can be registered from multiple locations at the same time and calls can be routed to all of its devices.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Use the following global configurations to modify this behavior:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>fs_multireg</span><span style='font-size:14.0pt;line-height:115%'>: -1: default, 0: no, 1: yes (enabled/disable multiple contact address)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:#984806'>allowforkforsignaling</span><span style='font-size:14.0pt;line-height:115%'>: 0: no,1: partial,2: yes,3: yes and remember old addresses (send sip request to multiple recipients at once)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373696"></a><a name="_Toc478281604"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Connecting with SIP clients</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can also connect SIP clients via the WebRTC-SIP proxy, although these are usually connected directly to your SIP server since there is no need for protocol conversion</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Register with two softphones and call from the first account to the second account.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Softphone configuration:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-bottom:0in;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><span style='font-size:14.0pt'>domain: your gateway IP or domain name (and the SIP port if your changed it from the standard 5060 UDP port)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;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><span style='font-size:14.0pt'>proxy: you can leave it empty</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;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><span style='font-size:14.0pt'>username / password: loaded from your SIP server (or from the Gateway if you created a special user with username/password authentication)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-bottom:0in;margin-bottom:.0001pt; line-height:normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Check “Client configuration” from the “Help” menu for the exact details.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>No other special settings are required (such as NAT, STUN, etc).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The network setting should be automatically handled by the server. If you don’t hear any voice you might change the RTP routing for the user(s) to “always route RTP” from MizuManage -> Users and devices -> Edit tab.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="node0" id="node0"></a><span style='font-size:14.0pt;line-height:115%; font-family:"Calibri",sans-serif'>VoIP Push notifications</span></h3> <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;line-height: normal'><a href="https://www.mizu-voip.com/Solutions/PushNotifications.aspx"><span style='font-size:14.0pt'>VoIP Push notifications</span></a><span style='font-size:14.0pt'> are useful to notify sleeping or closed mobile and web SIP clients on incoming call (wake-up and handle the incoming call) and instant messages (message display).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Both the Apple PushKit and the Google FCM are supported.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Set the following global config options (MManage -> Configurations form) to enable push notifications in your server or gateway:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-bottom:0in;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><span style='font-size:14.0pt'>pushnotification: 2</span></p> <p class="MsoListParagraphCxSpLast" style='margin-bottom:0in;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><span style='font-size:14.0pt'>pushnotification_chat: 2</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>If you are using the Mizu webphone or softphones, push notifications are supported by default. If you are using a third party SIP client or develop your own, then you need to add client support for it in your app to make it working. The process is described </span><a href="https://www.mizu-voip.com/Portals/0/Files/VOIP_Push_notifications.pdf"><span style='font-size:14.0pt'>here</span></a><span style='font-size:14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373697"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Webserver configuration</span></a></h3> <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%'>In case if you are using a web based WebRTC client, you will have to host it on your Web server such as IIS or Apache.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>However the WebRTC functionality has nothing to do with your Web server. Your Web server will be used only to serve your web content (including the html/css/JS needed for your WebRTC client), however once launched in a browser, WebRTC clients will connect directly to the WebRTC gateway (first with websocket, then also via ICE/DTLS/SRTP while in call).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Thus the WebRTC and <b>the WebRTC gateway have nothing to do with your Web server</b>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373698"></a><a name="_Toc478281605"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>STUN</span></a></h3> <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 WebRTC-SIP proxy has a built-in stun server <b>enabled by default</b>, usually listing on port 8080.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The stun server is listening on a port calculated at startup in the following way:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>-mainaportudp if that is set</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>-mainaport if that is set and if not 80</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>-otherwise on port 8090</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can easily get the STUN configuration required for your WebRTC clients from Help menu -> “Client configuration”.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The built-in stun service can be disabled by setting the<span style='color:#984806'> quickstun</span> to 0 and the <span style='color:#984806'>stunserver_ru</span>n to 0, however this is not recommended.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can also configure your WebRTC clients to use another STUN service, however this doesn’t make much sense as the built-in STUN in the MRTC Gateway is just perfect for WebRTC.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373699"></a><a name="_Toc478281606"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>TURN</span></a></h3> <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 WebRTC gateway has a built-in TURN service <b>enabled by default</b>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The TURN service is listening on the same port which you have set for websocket and API.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can configure this port from the Configuration Wizard -> Network page -> Access port</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>or by just changing the “<span style='color:#984806'>mainaport</span>” value in the global configuration (“Configuration” form).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>To change any turn related configuration, search for “<span style='color:#984806'>turn</span>” on the “Configuration” form. However, these should be rarely required because the default settings are just fine for all networks.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The Gateway also handles the TURN authentication automatically, allowing only the WebRTC clients which are successfully connected via your Gateway.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can easily get the TURN related configuration required for your WebRTC clients from Help menu -> “Client configuration”.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can also configure your WebRTC clients to use another TURN service, however this doesn’t make much sense as the built-in TURN in the MRTC Gateway is just perfect for WebRTC.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373700"></a><a name="_Toc478281608"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Codec</span></a></h3> <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 gateway has support for all the commonly used audio and video codecs:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>G.711, OPUS, G.729, G.723, GSM, iLBC, Speex, G.722, VP8, H.264</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The codec is negotiated automatically for each call depending on WebRTC client and your SIP server capabilities and other circumstances such as the available bandwidth.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The codec used for calls can be also influenced by the “<span style='color:#984806'>choosecodecs</span>” user configuration (which can be also set on the “Users and devices” form -> Functions page -> Allowed codec setting).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The codec to be available for the WebRTC endpoints can be set with the “fs_codec” option whose default value is PCMU,PCMA,OPUS,GSM,G722,VP8,H264. Usually there is no need to change this since a browser WebRTC client has support only for PCMU, PCMA and OPUS audio and VP8 or H264 video.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373701"></a><a name="_Toc478281609"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Transcoding</span></a></h3> <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%'>By default, the gateway will try to avoid transcoding when possible by negotiating a common codec between caller and called parties. If necessary, it will convert automatically from typical WebRTC codecs (OPUS and G.711) to typical telecom codecs (G.729, GSM and others). Change the following only if really needed.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If one of your peers has limited codec capabilities or the accepted codec(s) doesn’t match with the sender codecs, then set its “<span style='color:#984806'>needcodecconversion</span>” to 1 (for the target user which is usually a SIP Server user or Enduser). This can be also controlled on the "Functions" page of the "Users and devices" form.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Change the “<span style='color:#984806'>convertcodecs</span>” global config value to the target codec payload list. The default value is 0,8,18 which means PCMU,PCMA and G.729.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>To force always codec transcoding you can set the “fs_transcode” option to 2 (And set to 0 to avoid transcoding whenever possible).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The server is able to transcode between the following codecs: G.711 A-law , G.711 A-law ,G.729, G.723.1, GSM, OPUS, Speex 2,3,4,5,6 (narrowband, wideband and ultrawideband), G.726 and G.722. You might also have to set the “<span style='color:#984806'>choosecodecs</span>” field for the target user (same as the “convertcodecs” global config value) and the “convertcodecsforced” global config option to true.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Be aware that codec transcoding requires a high amount of CPU usage. For example one CPU (core) can handle around maximum 30-70 simultaneous transcodings between PCMU and G729 on full load.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373702"></a><a name="_Toc478281610"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>How to avoid transcoding?</span></a></h3> <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%'>Codec transcoding should be avoided whenever possible because it will increase the CPU usage and also will degrade the quality a bit. By default the MRTC gateway will try to negotiate a common codec between the endpoints and transcode only when strictly necessary.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The good news is that all common WebRTC clients (including browser WebRTC stacks) and SIP servers have G.711 (PCMU and PCMA) capabilities. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>To avoid codec transcoding, make sure to enable PCMU and/or PCMA on your SIP server.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If your SIP server has support for OPUS, then enable this also as its wideband variant has much better quality then G.711.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373703"></a><a name="_Toc478281611"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Optimal codec settings</span></a></h3> <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 MRTC WebRTC-SIP gateway is fine-tuned by default for best call quality.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>This means that when possible, it will use a wideband codec (usually OPUS as this is commonly supported by WebRTC and it has a very good quality, much better than PSTN). </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>However, OPUS doesn’t make sense if the call has to be routed to PSTN, because PSTN usually supports only narrowband codecs.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Usually WebRTC to WebRTC calls are done with OPUS.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If your SIP server has support for OPUS, then calls to SIP will also use this codec.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If a common wideband codec can’t be found, then it will use a narrowband codec (usually G.711 since this is supported by both SIP and WebRTC without the need for any transcoding).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If no common codec is found then it will transcode.</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%'>To achieve the best possible call quality, do the following:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Enable OPUS on your SIP server if it has support for this codec, especially if there are calls between users/extensions such as WebRTC to SIP or SIP to WebRTC calls. If all calls will go to landline or mobile (PSTN or a VoIP carrier, then OPUS is unnecessary, but it doesn’t hurt if it is enabled for occasional between users calls within your network)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>It is highly recommended to always enable G.711 (PCMU and/or PCMA). Actually this is enabled by default in every software, so just make sure to don’t disable it. Make sure that your carrier (if any) also has support for G.711, so the calls can pass end to end without the need for transcoding (most carriers allows G.711)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>If, for some reason you must send out a call (or receive calls) with other codecs (such as G.729) the MRTC Gateway will do the transcoding automatically, but this will have a cost in the performance and might degrade the call quality a bit.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Slow_call_setup"></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Slow call setup</span></h3> <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%'>With some WebRTC clients you might experience a considerable delay at call connect (call connecting/setup/in progress). This delay is usually caused by poorly written or configured WebRTC clients and it has nothing to do with the MRTC gateway. To solve the problem, check your WebRTC client configuration and set a timeout for ICE/TURN/TURN (or lover the existing timeout value). Contact the developers if it doesn’t have such configuration.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Around 4 second timeout should be enough in all circumstances (media candidates not collected within 4 seconds are likely to be invalid anyway).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373704"></a><a name="_Toc478281613"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Call recording</span></a></h3> <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%'>In MRTC versions released after 2018 Marc, you can easily control the chat recording from Tools menu -> Configure -> Recording -> Call.</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%'>The voice recording option can be set for any user by checking the “Voice Record” checkbox on the user configuration form in MManage (Users and Devices -> Functions tab). The followings fields will be affected in the tb_users table:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%;color:#984806'>record</span><span style='font-size:14.0pt;line-height:115%'>: 0 means no, 1 means yes</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%;color:#984806'>voicerecupload</span><span style='font-size:14.0pt;line-height:115%'>: you can specify a HTTP or FTP URL where the recorded files will be uploaded </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%'>Conversations will be saved in the directory specified by the “<span style='color:#984806'>serverftpvoice</span>” global config option.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The exact location will be: serverftpvoice\databasename\currentday\voice.xxx</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>A separate backup can be created in the directory specified by the “<span style='color:#984806'>voicebackupdir</span>” global config option.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Out of date recorded files can be deleted by setting the “<span style='color:#984806'>keeprecorded</span>” option accordingly (days to keep).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Recorded files are compressed and encrypted by default. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Recorded conversations can be played on the "CDR" form (Select the "Recorded Conversations" radio item, select the desired record and click on the Play button) or from the “Voice Record” form.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can also export the files as wav or mp3.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Users can replay the last record by the following DTMF digits: *4*</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%'>Call recording can make the routing and media path longer by disabling peer to peer routing and also will increase your server I/O and CPU usage.</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%'>To disable recording for all users, execute the following SQL’s in the “Direct Query” form:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>update [tb_users] set record = 0, voicerecupload=’’, RouteRTPCaller = 1, RouteRTPCalled = 1 where type in (0,5,9)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>Update tb_settings set valstr = '1' where keystr = 'defroutertp'</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] DROP CONSTRAINT [DF_tb_users_record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] ADD CONSTRAINT [DF_tb_users_record] DEFAULT 0 FOR [record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'> </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%'>If you wish to enable call recording for all users, execute the following SQL’s in the “Direct Query” form:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>update [tb_users] set record = 1, RouteRTPCaller = 7, RouteRTPCalled = 7 where type in (0,5,9)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>update tb_settings set valstr = '7</span> <span style='line-height:115%;color:#984806'>' where keystr = 'defroutertp'</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] DROP CONSTRAINT [DF_tb_users_record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] ADD CONSTRAINT [DF_tb_users_record] DEFAULT 1 FOR [record]</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%'>To force recording for all calls (even those where the media would be routed peer-to-peer otherwise), set the following global config options (This is usually not needed because even without these settings, calls between the WebRTC clients and your SIP servers will be recorded so these settings affects only WebRTC to WebRTC calls):</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>defroutertp=7 (force RTP routing by default)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>alloweuserusercalls=0 (optional setting to disable direct calls and route all calls via the upper SIP server –usually not needed)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>fs_directroutrfstofs= 1 (disable fast path routing)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>fs_forceoffrouteforvoicrec=2 (turn off webrtc direct routing. 0=no,1=yes,2=force for all calls)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>hasturn=0 (this will disable to TURN server as this doesn’t support call recording. TCP candidate will be still available if needed)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>disablep2prtprouting=1 (disable peer to peer media path detection)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>Delete the following routing rules if exists: check_user, check_user_notregistered, check_user_registered.</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'><i><span style='font-size:12.0pt;line-height:115%'>Note: MRTC versions released after 2017 November should automatically detect when recording is needed between two WebRTC endpoints and disable fast direct routing automatically.</span></i></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373705"></a><a name="_Toc478281612"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Chat recording</span></a></h3> <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 users IM history can be recorded and stored in server database, tb_messages table.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For this set the “<span style='color:#984806'>logmessenger</span>” global config option: 0=no,1=on low load,2=always</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You might also disable fast message forwarding by setting the “<span style='color:#984806'>fastmessagequeue</span>” configuration option to 0. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The recorded messages can be seen on the “Chat Logs” form MManage.</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%'>Note: In MRTC versions released after 2018 Marc, you can easily control the chat recording from Tools menu -> Configure -> Recording -> Chat.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373706"></a><a name="_Toc478281607"></a><a name="_First_test_calls"></a><a name="_DTMF"></a><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>DTMF</span></h3> <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 WebRTC-SIP Gateway will automatically handle DTMF sent from your client and server and it is capable also to transcode them.</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>By default will accept the DTMF digits as RFC 2833 or as SIP INFO methods from both your WebRTC clients and SIP servers.</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>By default it will send the DTMF digits as SIP INFO methods or as received</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%'>You just need to change the fs_dtmf_type_extern global config option to change the dtmf mode.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For this, open the MManage admin client -> Open the “Configurations” form (below the “Other” node in the tree view) -> enter “fs_dtmf_type_extern” in the search box -> Change the value after your needs -> Restart the gateway.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can restart the gateway from MManage Control menu or from windows services restart the mserver service.</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%'>Your SIP server -> MRTC Gateway -> WebRTC client:</span></u></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Both RFC2833 and SIP INFO are accepted from your gateway, so there is no reason to change this.</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%'>WebRTC client -> MRTC Gateway -> Your SIP server:</span></u></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>To set the DTMF method to use from our gateway to your SIP server, use the following settings:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For SIP INFO: set the fs_dtmf_type_extern to 1 (default)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For RFC2833: set the fs_dtmf_type_extern to 2</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>In-Band dtmf is not recommended.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can find all the DTMF related settings if you search for "dtmf" on the "Configurations" form.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373709"></a><a name="_Toc478281616"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Logs</span></a></h3> <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%'>You can find the Gateway logs from View menu -> Files -> Server log directory. Use a fast text reader to work with the files such as F3 from Total Commander.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The logs files are managed automatically by the gateway (auto deleting old log files or if you are low on disk-space).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can change the server log level from Control menu -> Log level.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373708"></a><a name="_Toc478281615"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Security</span></a></h3> <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 Gateway doesn’t store your VoIP user credentials (SIP password) so it is not a critical component to be protected from the security point of view.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Windows OS is as secure as any other (Linux/MAC/BSD) if you are using a clean install only to host the WebRTC gateway service with the built-in firewall turned on (but allowing the Gateway executables). This is because all Windows services are turned off on such server and kernel vulnerabilities exploitable via basic TCP/IP are extremely rare. If you are still concerned about Windows OS security, please read </span><a href="https://www.mizu-voip.com/Support/Blog/tabid/100/EntryID/8/Default.aspx"><span style='font-size:14.0pt;line-height:115%'>this blog post</span></a><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%'>If you are also running other software on the same box (such as a web server or any third party software) then please don’t blame our gateway security if your server gets hacked.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The WebRTC gateway is based on our Softswitch core which implements a long list of security measurements which are also applicable for the Gateway. See the details </span><a href="https://www.mizu-voip.com/Portals/0/Files/VoIP_Server_Security.pdf"><span style='font-size:14.0pt;line-height:115%'>here</span></a><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%'> </span></p> <h3 style='margin-top:0in'><a name="node1" id="node1"></a><span style='font-size:14.0pt;line-height:115%; font-family:"Calibri",sans-serif'>Feature support</span></h3> <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%'>Certain features (such as voice recording) don’t depend on any external support, however some features require support also by the WebRTC client and/or your SIP server (and/or SIP peers):</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For IM/chat to work, your WebRTC client and SIP server need to support the SIP MESSAGE standard which is the basic instant messaging protocol as described in </span><a href="https://www.ietf.org/rfc/rfc3428.txt"><span style='font-size:14.0pt; line-height:115%'>RFC 3428</span></a><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'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For DTMF to work your SIP server and WebRTC client must have support for </span><a href="https://www.ietf.org/rfc/rfc2833.txt"><span style='font-size:14.0pt; line-height:115%'>RFC 2833</span></a><span style='font-size:14.0pt;line-height: 115%'> or </span><a href="https://tools.ietf.org/html/rfc2976"><span style='font-size:14.0pt;line-height:115%'>SIP INFO</span></a><span style='font-size:14.0pt;line-height:115%'>. The gateway can also perform the conversion between these if needed. In-band DTMF conversion is not supported, however, it is routed as-is, so this method will also work if both your SIP server and WebRTC client use this method.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For presence to work your software needs support for PUBLISH (required/used only by WebRTC) and for SUBSCRIBE/NOTIFY as specified in </span><a href="https://tools.ietf.org/html/rfc3856"><span style='font-size:14.0pt; line-height:115%'>RFC 3856</span></a><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'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For video to work your SIP server and/or SIP peers need to support H.264 or VP8.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For call transfer you will need support for SIP REFER as described in </span><a href="https://tools.ietf.org/html/rfc3515"><span style='font-size:14.0pt; line-height:115%'>RFC 3515</span></a><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'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>For call forward you will need support SIP response code 301 (Moved Permanently) and/or 302 (Moved Temporarily).</span></p> <p class="MsoListParagraphCxSpLast" 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="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>If some feature is supported by your WebRTC client but not by your SIP server, then it will work only for WebRTC->WebRTC calls (and not for WebRTC->SIP or SIP->WebRTC).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The MRTC gateway also does its best to compensate on missing support. For example, the chat might work even without SIP MESSAGE support by carrying the message by SIP OPTIONS or SIP NOTIFY requests if your SIP server allows. Also the basic presence might work without SUBSCRIBE/NOTIFY support by using the peer registrar state.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc478373707"></a><a name="_Toc478281614"><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>Extra features</span></a></h3> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Class 5 features should be handled on your IP-PBX or Softswitch, however the Gateway also has some built-in features which might be useful for you (for example voicemail or call forward). Also for some functionalities there are no direct mapping between the WebRTC and SIP protocol so they can be solved only by extra features provided from the MRTC Gateway or on your Softswitch (for example conference). Other features don’t depend on the protocol used (no direct support in SIP or WebRTC) but must be implemented on the server side using some separate protocol (usually via a HTTP API exposed to clients).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Some extra PBX features can be enabled by setting the “<span style='color:#984806'>fs_pbx</span>” global configuration option to “1” (or disabled by setting the same to “0”).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Here are a few of the extra features:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>RTMP</span></b><span style='font-size:14.0pt'>: as a bonus, MRTC includes also a flash gateway module so it is compatible also with flash based voip clients. The RTMP is listening on port indicated by the “hasflash” setting defaulting to 1939. Disable by setting to 0.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Call forward</span></b><span style='font-size:14.0pt'>: can be enabled from users and devices form -> functions tab</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Call transfer</span></b><span style='font-size:14.0pt'>: available as specified SIP standards (via all devices with support for transfer)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Voicemail</span></b><span style='font-size:14.0pt'>: Enable/disable from the “<span style='color:#984806'>voicemail</span>“global config. Then enable/disable per user from “Users and devices” form -> “Functions” tab. Default access number is 5000 (with pin) and 5001 (with auto authentication). Auto-email forward is enabled by default.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Conference</span></b><span style='font-size:14.0pt'>: via SIP standard, via dtmf *1*number# or via conference rooms using extensions between 5100-5199 for narrowband and 5200-5299 for wideband. You can use the “getconfroom” API to auto allocate a conference room and its access number for the user.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Special numbers</span></b><span style='font-size:14.0pt'> and IVR’s such as music, record/playback, vide record/playback and others</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Missed call notifications</span></b><span style='font-size:14.0pt'> by email</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Barge in: </span></b><span style='font-size:14.0pt'>via the “Voice here” form, or right click on current call or 5009 access number</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Softphones</span></b><span style='font-size:14.0pt'>: Mizutech can also offer customized/branded softphones which works with the Gateway or directly with your SIP server: </span><a href="https://www.mizu-voip.com/Software/WebPhone.aspx"><span style='font-size: 14.0pt'>Browser webphone</span></a><span style='font-size:14.0pt'>, </span><a href="https://www.mizu-voip.com/Software/Softphones/Windowssoftphone.aspx"><span style='font-size:14.0pt'>Windows softphone</span></a><span style='font-size: 14.0pt'>, </span><a href="https://www.mizu-voip.com/Software/Softphones/AndroidSoftphone.aspx"><span style='font-size:14.0pt'>Android softphone</span></a><span style='font-size: 14.0pt'>, </span><a href="https://www.mizu-voip.com/Software/Softphones/iPhoneSoftphone.aspx"><span style='font-size:14.0pt'>iOS softphone</span></a><span style='font-size:14.0pt'>, </span><a href="https://www.mizu-voip.com/Software/Softphones/SymbianSoftphone.aspx"><span style='font-size:14.0pt'>Symbian dialer</span></a><span style='font-size:14.0pt'>, </span><a href="https://www.mizu-voip.com/Software/Softphones.aspx"><span style='font-size:14.0pt'>Other softphones</span></a></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>IVR</span></b><span style='font-size: 14.0pt'>: The IVR module can be used for various tasks like access numbers, callback, customer support etc.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>You can assign different IVR’s to different access numbers by using the “<b>Campaigns</b>” form. To create a new campaign, just click on the + sign and enter a “name” for the new record. The most important configuration for an IVR campaign is the script. Switch to the “details” tab to select a “Script”.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Scripts can be created by using the “IVR” form. The gateway is shipped with several preconfigured script examples, but you should easily add new scripts or modify the existing ones by following the </span><a href="https://www.mizu-voip.com/LinkClick.aspx?fileticket=iP2DA5gwh-s%3d&tabid=176&mid=675"><span style='font-size:14.0pt'>admin guide</span></a><span style='font-size:14.0pt'> or the </span><a href="https://www.mizu-voip.com/Portals/0/Files/IVR.pdf"><span style='font-size:14.0pt'>IVR documentation</span></a><span style='font-size: 14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>SMS: </span></b><span style='font-size:14.0pt'>by default the gateway is capable to route SMS message as SIP MESSAGE (RFC 3428). However, it is also possible to use an external service which can be accessed by a HTTP API. Just set the “<span style='color:#984806'>smsurl</span>” smsurl global config option or create SMS GW endpoints if you wish to use more than one provider. A guide can be found </span><a href="http://www.mizu-voip.com/Portals/0/Files/sms_softswitch.pdf"><span style='font-size:14.0pt'>here</span></a><span style='font-size:14.0pt'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>API</span></b><span style='font-size: 14.0pt'>: the Gateway also has a HTTP API which can be used for various tasks. See the </span><a href="https://www.mizu-voip.com/Portals/0/Files/VoIP_Server_API.pdf"><span style='font-size:14.0pt'>API documentation</span></a><span style='font-size: 14.0pt'> for the details.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><b><span style='font-size:14.0pt;line-height:115%'>Many others</span></b><span style='font-size:14.0pt;line-height:115%'> features are enabled for you by default and some of them can be changed/fine-tuned from configurations, such as caller-id, ring groups, call hold, call waiting/park/pickup and others. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The WebRTC-SIP proxy is based on the VoIP Server core thus you can also access almost all of the Softswitch features. See the </span><a href="https://www.mizu-voip.com/Portals/0/Files/MAdminGuide.pdf"><span style='font-size:14.0pt;line-height:115%'>Softswitch documentation</span></a><span style='font-size:14.0pt;line-height:115%'> for the details. </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%'>If you are not sure where to find a specific configuration option, search for your keyword in:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-bottom:0in;margin-bottom:.0001pt; text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>“Configurations” form within MManage</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;margin-bottom: .0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>This guide</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;margin-bottom: .0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><a href="https://www.mizu-voip.com/Portals/0/Files/MAdminGuide.pdf"><span style='font-size:14.0pt;line-height:115%'>VoIP server guide</span></a><span style='font-size:14.0pt;line-height:115%'> (most of the settings can be applied also for the MRTC gateway)</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;margin-bottom: .0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>4.<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/Support/Documentations.aspx"><span style='font-size:14.0pt;line-height:115%'>other documentations</span></a><span style='font-size:14.0pt;line-height:115%'> (some of them are relevant also for MRTC)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-bottom:0in;margin-bottom:.0001pt; text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>5.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Or ask </span><a href="mailto:info@mizu-voip.com?subject=WebRTC-SIP%20Gateway%20support%20request"><span style='font-size:14.0pt;line-height:115%'>Mizutech support</span></a></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h1 style='margin-top:0in'><a name="node2" id="node2"></a><span style='font-size:16.0pt;line-height:115%; font-family:"Calibri",sans-serif'>FAQ</span></h1> <p class="MsoNormal"> </p> <h3 style='margin-top:0in'><a name="node3" id="node3"></a><span style='font-size:14.0pt;line-height:115%'>How to get my WebRTC-SIP gateway</span></h3> <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%'>Follow these steps:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Have a look at the </span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP.aspx"><span style='font-size:14.0pt;line-height:115%'>home page</span></a><span style='font-size:14.0pt;line-height:115%'> and eventually download and try the free </span><a href="https://www.mizu-voip.com/G/download/MRTC_Setup.exe"><span style='font-size:14.0pt;line-height:115%'>WebRTC-SIP gateway</span></a><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'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>If seems to be ok for your needs, purchase from </span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP/WebRTC-SIP-Gateway-Pricing.aspx"><span style='font-size:14.0pt;line-height:115%'>here</span></a><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'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>On your payment Mizutech will send the download link by email (you can just reinstall to upgrade any existing versions without data/settings lost)</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>4.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Contact </span><a href="mailto:info@mizu-voip.com?subject=WebRTC-SIP%20gateway%20support%20request"><span style='font-size:14.0pt;line-height:115%'>mizutech support</span></a><span style='font-size:14.0pt;line-height:115%'> for any help or if you run into any issue using the MRTC gateway.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'> </span></p> <h3><a name="node4" id="node4"></a><span style='font-size:14.0pt;line-height:115%'>Can’t connect/register</span></h3> <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'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Don’t use special UTF characters in username/password (some SIP servers don’t like non-ASCII characters).</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Make sure that you are using a correct SIP username/password valid on your SIP server.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Some server require to set a different username/auth username (Username is usually the Extension ID). If you are using the Mizu WebPhone, then set the Extension ID as the webphone “Caller ID” (“username” parameter) and set the Auth user name as the webphone “Username” (“sipusername” parameter).</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>4.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check if otherwise you can connect with a simple SIP client (first directly to your server, second to the gateway). If this works, then the problem is WebRTC or WebSocket related.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>5.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Test if the gateway WebRTC service is listening from MManage -> Tools menu -> Configure -> Tests.</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>6.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check the browser console logs and the gateway logs (MManage -> View menu -> Files -> Server logs).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3><a name="node5" id="node5"></a><span style='font-size:14.0pt;line-height:115%'>Can’t make calls</span></h3> <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'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Make a test call first from a simple SIP client such as mizu softphone or x-lite (first directly to your server, second to the gateway). If this works, then the problem is WebRTC or WebSocket related.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check the CDR disconnect reason (if the call reached the gateway) from MMAnage -> CDR form.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Make sure to allow Media Permissions for WebRTC in your browser.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>4.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check the browser console output.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>5.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check if the call reaches the gateway and inspect the gateway logs (MManage -> View menu -> Files -> Server logs).</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>6.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check if the call reaches your SIP server and inspect your server logs.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3><a name="node6" id="node6"></a><span style='font-size:14.0pt;line-height:115%'>Incoming calls not received</span></h3> <p class="MsoNormal"> </p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check if the call reaches your SIP server. If not, then the problem has nothing to do with WebRTC.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check if the call reaches the WebRTC gateway. In not, then make sure that the WebRTC client is registered to your server at the time when the inbound call is routed to it.</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Check if the call reaches the WebRTC client (check the browser console output with detailed trace/log level, search for “INVITE sip”).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3><a name="node7" id="node7"></a><span style='font-size:14.0pt;line-height:115%'>Calls are disconnecting</span></h3> <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; line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family: Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Call disconnection immediately upon setup can have many reasons such as codec incompatibility issues, NAT issues, DTLS/SRTP setup problems or audio problems.</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family: Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Call disconnection after 20 seconds might be caused by MEDIATIMEOUT and MEDIATIMEOUTSTART global config option if there is no audio (RTP stream) or one way audio only.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3><a name="node8" id="node8"></a><span style='font-size:14.0pt;line-height:115%'>No voice or one side audio</span></h3> <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%'>Change the RTP media routing for one or all users.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For one user this can be changed from the “Users and Devices” form -> Functions page -> Media relay.</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%'>For all users it can be changed in the following ways:</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%'>To disable media routing, execute the following SQL’s in the “Direct Query” form:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>update [tb_users] set RouteRTPCaller = 1, RouteRTPCalled = 1 where type in (0,5,9)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>Update tb_settings set valstr = '1' where keystr = 'defroutertp</span> <span style='line-height:115%;color:#984806'>'</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] DROP CONSTRAINT [DF_tb_users_record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] ADD CONSTRAINT [DF_tb_users_record] DEFAULT 0 FOR [record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'> </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%'>To enable media routing, execute the following SQL’s in the “Direct Query” form:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>update [tb_users] set RouteRTPCaller = 7, RouteRTPCalled = 7 where type in (0,5,9)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>update tb_settings set valstr = '7</span> <span style='line-height:115%;color:#984806'>' where keystr = 'defroutertp'</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] DROP CONSTRAINT [DF_tb_users_record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:gray'>ALTER TABLE [tb_users] ADD CONSTRAINT [DF_tb_users_record] DEFAULT 1 FOR [record]</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%'>To force the media routing even for WebRTC-WebRTC calls, also set the following global config options:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>defroutertp=7 (force RTP routing by default)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>alloweuserusercalls=0 (optional setting to disable direct calls and route all calls via the SIP server –usually not needed)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>fs_directroutrfstofs=1 (disable fast path routing)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'>fs_forceoffrouteforvoicrec=2 (turn off webrtc direct routing. 0=no,1=yes,2=force for all calls)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%;color:#984806'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="node9" id="node9"></a><span style='font-size:14.0pt;line-height:115%'>Long call setup time</span></h3> <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%'>If you experience long setup or ring time, that is almost always due to poor client side WebRTC implementation. Make sure to use a WebRTC client with proper ICE timeout limitations such as the mizu webphone. It is normal for ICE queries to take 20 or more seconds, however it is not normal to wait for these to complete on the client side as such delayed responses are always useless and mostly caused by unused network interfaces.</span></p> <h3><a name="node10" id="node10"></a><span style='font-size:14.0pt;line-height:115%'>Chat doesn’t work</span></h3> <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%'>Make sure that your SIP server and WebRTC client has support for SIP MESSAGE as described in </span><a href="https://www.ietf.org/rfc/rfc3428.txt"><span style='font-size:14.0pt; line-height:115%'>RFC3428</span></a><span style='font-size:14.0pt;line-height: 115%'>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>Most old Asterisk installations might not have support for this </span><a href="http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging"><span style='font-size:14.0pt'>by default</span></a><span style='font-size:14.0pt'>. You might use </span><a href="http://www.kamailio.org/"><span style='font-size: 14.0pt'>Kamailio</span></a><span style='font-size:14.0pt'> for this purpose or any other </span><a href="https://www.mizu-voip.com/Products/VoIPServer.aspx"><span style='font-size:14.0pt'>softswitch</span></a><span style='font-size:14.0pt'> (most of them has support for RFC 3428).</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For new versions of Asterisk based SIP servers you need to properly </span><a href="http://stackoverflow.com/a/23110898/3376017"><span style='font-size:14.0pt'>configure SIP MESSAGE routing</span></a><span style='font-size:14.0pt'> in order for IM to work.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>See also </span><a href="http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging"><span style='font-size:14.0pt'>Asterisk patch</span></a><span style='font-size:14.0pt'> or </span><a href="http://issues.freepbx.org/browse/FREEPBX-7058"><span style='font-size:14.0pt'>FreePBX settings</span></a><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%'> </span></p> <h3><a name="node11" id="node11"></a><span style='font-size:14.0pt;line-height:115%'>Presence doesn’t work</span></h3> <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;line-height: normal'><span style='font-size:14.0pt'>Presence is implemented as standard SIP PUBLISH/SUBSCRIBE/NOTIFY so your SIP server must support </span><a href="https://www.ietf.org/rfc/rfc3265.txt"><span style='font-size:14.0pt'>RFC 3265</span></a><span style='font-size:14.0pt'> and </span><a href="https://www.ietf.org/rfc/rfc3856.txt"><span style='font-size:14.0pt'>RFC 3856</span></a><span style='font-size:14.0pt'> in order for this to work.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For Asterisk based servers you can find configuration details </span><a href="https://www.voip-info.org/wiki/view/Asterisk+presence"><span style='font-size:14.0pt'>here</span></a><span style='font-size:14.0pt'> and </span><a href="https://wiki.asterisk.org/wiki/display/AST/Configuring+chan_sip+for+Presence+Subscriptions"><span style='font-size:14.0pt'>here</span></a><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%'> </span></p> <h3><a name="node12" id="node12"></a><span style='font-size:14.0pt;line-height:115%'>Video doesn’t work</span></h3> <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%'>Make sure that your server has full support for H.264 and VP8 or it bypass the media routing and forwards the SDP attributes correctly.</span></p> <h3><a name="node13" id="node13"></a><span style='font-size:14.0pt;line-height:115%'>DTMF doesn’t work</span></h3> <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%'>WebRTC clients are usually sending the DTMF with SIP INFO so make sure that your SIP server can handle this DTMF method.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Adjust the gateway global configuration after your needs:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-bottom:0in;margin-bottom:.0001pt; text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family: Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>fs_dtmf_type_intern: //dtmf-type to clients : 0=default, 1=info, 2=rfc2833</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;margin-bottom: .0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%; font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>fs_dtmf_type_extern: dtmf-type to servers: 0=default, 1=info, 2=rfc2833</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-bottom:0in;margin-bottom: .0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%; font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>fs_dtmf_duration: dtmf-duration. def: 600</span></p> <p class="MsoListParagraphCxSpLast" style='margin-bottom:0in;margin-bottom:.0001pt; text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family: Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>for other DTMF related settings, search for “dtmf” in the global configuration</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>For more details, read </span><a href="start.htm#_DTMF"><span style='font-size:14.0pt;line-height:115%'>here</span></a><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%'> </span></p> <h3><a name="node14" id="node14"></a><span style='font-size:14.0pt;line-height:115%'>I don’t see some calls on my SIP server</span></h3> <p class="MsoNormal"><span style='font-size:14.0pt;line-height:115%'><br> </span><span style='font-size:14.0pt;line-height:115%'>The WebRTC gateway by default will route the calls between the endusers bypassing your SIP server with the advantage of freeing up resource utilizations on your SIP server.</span></p> <p class="MsoNormal"><span style='font-size:14.0pt;line-height:115%'>You can disable this behavior by setting the <span style='color:#C0504D'>alloweuserusercalls</span> and <span style='color:#C0504D'>fs_directroutrfstofs</span></span></p> <p class="MsoNormal"><span style='font-size:14.0pt;line-height:115%'>global config options to <span style='color:#C0504D'>0</span> thus forcing all calls to be routed via your server.</span></p> <h3><a name="node15" id="node15"></a><span style='font-size:14.0pt;line-height:115%'>Call quality problems</span></h3> <p class="MsoNormal"><span style='font-size:14.0pt;line-height:115%'> </span></p> <p class="MsoNormal"><span style='font-size:14.0pt;line-height:115%'>Call quality is influenced primarily by the followings:</span></p> <p class="MsoListParagraphCxSpFirst" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Network conditions (QoS)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Codec used to carry the media (wideband has better quality)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>AEC, AGC and denoise (enable/disable these in your client after your needs)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Hardware: enough CPU power and quality headset/microphone/speaker (try a headset, try on another device)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>OS/Browser (drivers problems)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Settings (codec, jitter size, AEC, etc)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Software problem or bug</span></p> <p class="MsoListParagraphCxSpLast" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>VoIP route (carrier/call termination/voip service provider call quality)</span></p> <p class="MsoNormal"><span style='font-size:14.0pt;line-height:115%'>If you have call quality issues then the followings should be verified:</span></p> <p class="MsoListParagraphCxSpFirst" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Check your network quality: speed, packet loss, delay and jitter </span><a href="https://sourceforge.net/speedtest/"><span style='font-size:14.0pt; line-height:120%'>here</span></a><span style='font-size:14.0pt;line-height: 120%'>. (Other tools: </span><a href="http://speedtestatl.cbeyond.net/index.html"><span style='font-size:14.0pt; line-height:120%'>a</span></a><span style='font-size:14.0pt;line-height:120%'> , </span><a href="http://myspeed.visualware.com/"><span style='font-size:14.0pt; line-height:120%'>b</span></a><span style='font-size:14.0pt;line-height:120%'>, </span><a href="http://www.speedtest.net/"><span style='font-size:14.0pt; line-height:120%'>c</span></a><span style='font-size:14.0pt;line-height:120%'>, </span><a href="http://www.pingtest.net/"><span style='font-size:14.0pt; line-height:120%'>d</span></a><span style='font-size:14.0pt;line-height:120%'>)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Check whether you have good call quality using a third party softphone from the same location (try X-Lite for example). If not, than the problem is not related to the gateway (continue to look for server, termination gateway or bandwidth issues)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Make sure that the CPU load is not near 100% when you are doing the tests </span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Make sure that you have enough </span><a href="https://www.bandcalc.com/"><span style='font-size:14.0pt;line-height:120%'>bandwidth</span></a><span style='font-size:14.0pt;line-height:120%'>/QoS for the </span><a href="start.htm#_Toc478373700"><span style='font-size:14.0pt;line-height:120%'>codec</span></a><span style='font-size:14.0pt;line-height:120%'> that you are using (G.729 requires around 24 kbits. G.711 requires around 80 kbits. Opus requires around 20-50 kbits. However the overall bandwidth is rarely the problem and you should check if you have these minimums constantly, with less packet loss then 2% and no big variations in roundtrip delays)</span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Change the codec </span></p> <p class="MsoListParagraphCxSpMiddle" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Try to set only one single codec (such as PCMU) to make sure that the problem is not caused by poor server side handling of the automatically selected codec</span></p> <p class="MsoListParagraphCxSpLast" style='text-indent:-.25in;line-height:120%'><span style='font-size:14.0pt;line-height:120%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:120%'>Perform a network trace (With </span><a href="https://www.wireshark.org/download.html"><span style='font-size:14.0pt;line-height:120%'>Wireshark</span></a><span style='font-size:14.0pt;line-height:120%'> for example; Filter for SIP and RTP. Check missing or duplicated packets.)</span></p> <h3><a name="node16" id="node16"></a><span style='font-size:14.0pt;line-height:115%'>How to upgrade?</span></h3> <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%'>To upgrade or update to the latest version, all you have to do is to run the installer. This will overwrite your old binaries while keeping all your existing data intact, so you will not lose any settings. On first startup it will automatically upgrade also the data structures, so there is no need for any manual action.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h1 style='margin-top:0in'><a name="_Toc478373711"><span style='font-size:16.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Resources</span></a></h1> <p class="MsoListParagraphCxSpFirst" 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="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.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/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> </div>
Contents
Topics
Search
About
Features
How It Works
Legal Agreement
Setup and configuration
Install
Configuration Wizard
SIP Server Configuration
WebRTC Client Configuration
Troubleshooting
Management and Monitoring
MizuManage
User Management
Add SIP Server
Add User
Outbound Routing
Inbound Routing
CDR
Statistics and Monitoring
Start/Stop
Backup/Restore
Additional Settings
Settings
Global Configuration
Clean Install
Domain name
SSL certificate
Change IP address
Ports
Gateway behind NAT
Multiple SIP servers
Calls between endusers/extensions
Multiple registrations and call fork
Connecting with SIP clients
VoIP Push notifications
Webserver configuration
STUN
TURN
Codec
Transcoding
How to avoid transcoding?
Optimal codec settings
Slow call setup
Call recording
Chat recording
DTMF
Logs
Security
Feature support
Extra features
FAQ
How to get my WebRTC-SIP gateway
Can’t connect/register
Can’t make calls
Incoming calls not received
Calls are disconnecting
No voice or one side audio
Long call setup time
Chat doesn’t work
Presence doesn’t work
Video doesn’t work
DTMF doesn’t work
I don’t see some calls on my SIP server
Call quality problems
How to upgrade?
Resources
About
Add SIP Server
Add User
Additional Settings
Backup/Restore
Call recording
Call quality problems
Calls are disconnecting
Calls between endusers/extensions
Can’t connect/register
Can’t make calls
CDR
Change IP address
Chat recording
Chat doesn’t work
Clean Install
Codec
Configuration Wizard
Connecting with SIP clients
Domain name
DTMF
DTMF doesn’t work
Extra features
FAQ
Feature support
Features
Gateway behind NAT
Global Configuration
How to avoid transcoding?
How to get my WebRTC-SIP gateway
How It Works
How to upgrade?
I don’t see some calls on my SIP server
Inbound Routing
Incoming calls not received
Install
Legal Agreement
Logs
Long call setup time
Management and Monitoring
MizuManage
Multiple SIP servers
Multiple registrations and call fork
No voice or one side audio
Optimal codec settings
Outbound Routing
Ports
Presence doesn’t work
Resources
Security
Settings
Setup and configuration
SIP Server Configuration
Slow call setup
SSL certificate
Start/Stop
Statistics and Monitoring
STUN
Transcoding
Troubleshooting
TURN
User Management
Video doesn’t work
VoIP Push notifications
WebRTC Client Configuration
Webserver configuration
WebRTC-SIP Proxy
