SIP SBC
<!-- 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="SIP_SBC_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="MsoToc3"><a href="start.htm#_Toc103673763">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#_Toc103673764">Quick setup<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#_Toc103673765">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#_Toc103673766">Requirements<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#_Toc103673767">Install<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#_Toc103673768">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#_Toc103673769">Troubleshooting<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#_Toc103673770">MizuManage<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="MsoToc3"><a href="start.htm#_Toc103673771">User Management<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#_Toc103673772">SIP Servers<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#_Toc103673773">SIP Clients and Trunks<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#_Toc103673774">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#_Toc103673775">Inbound Routing<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#_Toc103673776">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#_Toc103673777">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#_Toc103673778">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#_Toc103673779">Upgrade<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#_Toc103673780">Backup/Restore<span style='color: windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>16</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673781">Settings<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>16</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673782">Global Configuration<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>16</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673783">Change SBC IP address<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>17</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673784">Change SIP server address<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>17</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673785">Ports<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>19</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673786">RTP Routing<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>20</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673787">SBC behind NAT<span style='color: windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>21</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673788">Calls between endusers/extensions<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>22</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673789">Multiple registrations and call fork<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>22</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673790">Codec<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>22</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673791">Transcoding<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>23</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673792">Call recording<span style='color: windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>24</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673793">Chat recording<span style='color: windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>25</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673794">WebRTC<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>25</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673795">Push notifications<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>25</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673796">API<span style='color:windowtext; display:none;text-decoration:none'> </span><span style='color:windowtext;display:none;text-decoration:none'>26</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673797">Logs<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>26</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673798">Security<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>26</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673799">High availability<span style='color:windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>27</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673800">Extra features<span style='color: windowtext;display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>27</span></a></p> <p class="MsoToc3"><a href="start.htm#_Toc103673801">Resources<span style='color:windowtext; display:none;text-decoration:none'>. </span><span style='color:windowtext;display:none;text-decoration:none'>29</span></a></p> <p class="MsoNormal"> </p> <h3><a name="_Toc103673763"><span style='font-size:14.0pt;line-height:115%'> </span></a></h3> <h3><a name="node0" id="node0"></a><span style='font-size:14.0pt;line-height:115%'>About</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 Mizu </span><a href="https://www.mizu-voip.com/Software/SIPSBC.aspx"><span style='font-size: 14.0pt;line-height:115%'>SIP SBC</span></a><span style='font-size:14.0pt; line-height:115%'> is an easy to use software solution to control SIP signaling and media streams capable to enforce security and to perform various tasks such as validation of SIP sessions and NAT handling.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The SIP SBC can be installed on Windows operating systems and runs as an NT service.</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="_Toc103673764"><span style='font-size:14.0pt;line-height:115%'>Quick setup</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%'>Usage instructions in short:</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%'>Download the installer (the free/demo version can be downloaded from </span><a href="https://www.mizu-voip.com/Portals/0/Files/MPUSH_Setup.exe"><span style='font-size:14.0pt;line-height:115%'>here</span></a><span style='font-size:14.0pt;line-height:115%'>) and install it on a Windows machine (you can also use a virtual server or VPS)</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%'>Launch the MManage admin client (might be launched automatically by the installer) and follow the Configuration Wizard (can be found from the Config menu if not started 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%'>Take special attention at the Network page (configure with the correct IP address) and SIP Server page (enter the correct details about your SIP server, you will be able to add more later) </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%'>Enable the main executables on the Windows firewall: mserver.exe, mserverftp.exe, pushkit.exe, lego.exe.</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%'>If you are hosting the SBC behind NAT and peers will be located on outside network (like softphones connecting from the public internet), then configure port forwarding on your external router or NAT device for the ports listed at Config -> Network -> Active ports.</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%'>For a quick test, configure two softphones (use the gateway domain or IP:port as the SIP domain or outbound proxy setting) and make a test call.</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="_Toc103673765"><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'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><b><span style='font-size:14.0pt;line-height:115%'>GUI</span></b><span style='font-size:14.0pt;line-height:115%'>: comfortable graphical user interface for configuration, management and monitoring (real-time, CDR's, 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'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><b><span style='font-size:14.0pt;line-height:115%'>NAT</span></b><span style='font-size:14.0pt;line-height:115%'>: the SBC has built-in NAT traversal, support capable to gracefully handle all types of NAT's, firewalls and routers</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><b><span style='font-size:14.0pt;line-height:115%'>Security</span></b><span style='font-size:14.0pt;line-height:115%'>: DoS and DDoS attack protection with intelligent firewall, topology-hiding, access control, call limits, rate limiter, stream and session protection, blacklist, fraud call detection and various other thresholds </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><b><span style='font-size:14.0pt;line-height:115%'>Internetworking</span></b><span style='font-size:14.0pt;line-height:115%'>: compatible with all SIP devices, will resolve any incompatibility issues between 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'><span style='font-size:14.0pt;line-height:115%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><b><span style='font-size:14.0pt;line-height:115%'>Routing</span></b><span style='font-size:14.0pt;line-height:115%'>: intelligent routing of calls can be done by various rules such as priority, quality and load-balancing</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><b><span style='font-size:14.0pt;line-height:115%'>Failover</span></b><span style='font-size:14.0pt;line-height:115%'>: it can detect failed servers and reroute the calls to others or automatically temporarily disable servers </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><b><span style='font-size:14.0pt;line-height:115%'>RTP relay</span></b><span style='font-size:14.0pt;line-height:115%'>: bypass, route or off-route the media with various enhancements such as NAT handling, QoS and stream control. It can be set to automatic (will route the media only if necessary) or enforced by manual 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><b><span style='font-size:14.0pt;line-height:115%'>Transcoding</span></b><span style='font-size:14.0pt;line-height:115%'>: transcode between various codec's such as Opus, Speex, G.729, G.723, G.711 (PCMU/PCMA), GSM, iLBC (this is done automatically by default only when necessary and can be overridden manually)</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><b><span style='font-size:14.0pt;line-height:115%'>Optional modules</span></b><span style='font-size:14.0pt;line-height:115%'>: billing, user management, registrar, class 5 features, WebRTC-SIP, push notifications, tunneling, high-speed load balancer (up to 2 000 000 simultaneous 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'> </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> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3><a name="_Toc103673766"><span style='font-size:14.0pt;line-height:115%'>Requirements</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 mizu SIP SBC can be installed on any server or PC running Windows OS.</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%'>OS: Windows Server 2003/2008/2012/2016/2019/2022 (Windows XP/Vista/7/8/10/11 for home usage or testing). Both 32 and 64 bit versions are supported. You can use the cheap Web edition license. </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%'>CPU: a cheap dual core cpu for less than 500 parallel calls, a best-buy 4+ core CPU for less than 2000 parallel calls or a high-end CPU for more. </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%'>RAM: 1 GB RAM for less than 500 simultaneous calls, 6 GB RAM for less than 2000 simultaneous calls or 8+ GB for more.</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%'>Disk: 60 GB HDD (or more if your traffic is high and wish to store all CDR's or more logs or voice recording)</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%'>Example hardware for 30000 users with 2000 simultaneous calls: </span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.0in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>a best buy 4-8 core AMD Epyc or Intel Xeon such as Xeon E5-2665 which costs only ~$90</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.0in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>8 GB RAM</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.0in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>80 GB HDD</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="MsoNormalCxSpMiddle" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:black'>We recommend to host the SBC service on a server with a public static IP, using standard </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%;color:black'> (80 for HTTP, 443 for HTTPS/TLS, 5060 for SIP, 5061 for SIPS).</span></p> <p class="MsoNormalCxSpMiddle" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:black'>In case if some other app on your machine is already listening on these ports then you should stop the app if unnecessary or otherwise configure your gateway to bind to another IP (if you have multiple IP addresses) or otherwise configure other ports to be used by the SIP SBC.</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="_Toc103673767"><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 SIP SBC has a standard easy to use installer.</span></p> <p class="MsoNormalCxSpMiddle" style='margin-bottom:0in;margin-bottom:.0001pt'><i><span style='font-size:12.0pt;line-height:115%;color:black'>Quick tip before to start the installer: Windows servers might have IIS running by default. Stop it if not needed (World Wide Web Publishing Service from Services).</span></i></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%'>Download the SIP SBC installer from </span><a href="https://www.mizu-voip.com/Portals/0/Files/SIP_SBC_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%'>2.<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%'>3.<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="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%'> </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="_Toc103673768"></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 “Config” menu.</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 SBC (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 SBC ports or the third party app port or bind them to separate IP 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'>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 SBC 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 SIP clients or trunks 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'><span style='font-size:14.0pt;line-height:115%'>Once the SIP SBC is installed and configured correctly, there is nothing much to manage 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;line-height: normal'><a name="_Toc285322899"><span style='font-size:14.0pt'> </span></a></p> <h3 style='margin-top:0in'><a name="_Toc103673769"><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'>Look for any issues from MizuManage:</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<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:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<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:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<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:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<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'>Make sure that you have basic connectivity to the important ports such as SIP signaling 5060 and RTP ports</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 SBC 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 or there is one way audio then you might change the RTP routing for the user(s) to “always route RTP” from MizuManage -> Users and devices -> Edit tab or globally from Config menu -> Configurations -> RTP Relay -> Route.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Check if there is any firewall blocking the required ports and if your NAT device/router/firewall correctly forwards the required ports (Config menu -> Network -> Active ports)</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 -> Files Menu -> Folders -> Server logs.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span>In case if you are using push, then you can find the register from the users by searching for the followings in the log:</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-family:Wingdings'>§<span style='font:7.0pt "Times New Roman"'> </span></span>EVENT,creating new registrar server with usr: USERNAME</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-family:Wingdings'>§<span style='font:7.0pt "Times New Roman"'> </span></span>EVENT,reuse registrar endpoint for USERNAME with callid X</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-family:Wingdings'>§<span style='font:7.0pt "Times New Roman"'> </span></span>EVENT, (fast) new reg userobj from IP as reg from USERNAME</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-family:Wingdings'>§<span style='font:7.0pt "Times New Roman"'> </span></span>EVENT, (fast) reg send to USERNAME Welcome back</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:1.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-family:Wingdings'>§<span style='font:7.0pt "Times New Roman"'> </span></span>EVENT,force unreg from upper server USERNAME</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Open the last </span><a href="start.htm#_Logs"><span style='font-size:14.0pt'>log</span></a><span style='font-size:14.0pt'> 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 </span><a href="sip:callednumber"><span style='font-size:14.0pt'>sip:callednumber</span></a><span style='font-size:14.0pt'>”.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<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:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Use some network capture tool (such as </span><a href="https://www.wireshark.org/"><span style='font-size:14.0pt'>Wireshark</span></a><span style='font-size:14.0pt'>) for further troubleshooting.</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=SIP%20SBC%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> <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="_Toc103673770"><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 SIP SBC 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 SBC, you will still need to use your Softswitch/PBX to manage your Users (authentication, authorization, and accounting), Routing 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 SBC also provides you complete monitoring and GUI based configuration managements, so you can easily change settings or find connectivity issues.</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 below the quick filter or from the Fields 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 the Config menu). Use this to (re)configure your SBC’s 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="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%'>From the “Fields” menu you can manipulate the selected dataset (grids, etc).</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 most important settings can be configured from Config menu -> 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%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc103673771"><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 SBC 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 SBC!</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 SBC internals, auto created and managed by the SBC. Except your extension usernames, no authentication credentials (password) or any sensitive information (such as billing) are stored on the SIP SBC. 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 SBC might have information only about their username and display number (the SBC 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 SBC (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> global config value 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="_Toc103673772"></a><a name="_Add_SIP_Server"></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>SIP Servers</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'><span style='font-size:14.0pt;line-height:115%'>A default/first SIP server record is automatically created for you once you complete the Configuration Wizard. Use the “Add Upper SIP server” form (available from “Config” menu -> Users -> Quick Add) in case if you have more SIP servers where the traffic from the clients’ needs to be sent or accepted from. (Or just create Traffic sender + SIP Server + add the SIP server to the Routing).</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. <br> Also don’t forget to add the SIP server to the routing (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'><b><span style='font-size:14.0pt'>Add SIP Server:</span></b></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 Config menu -> Users -> Quick Add)</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'>To add a SIP server open the “Add Upper SIP server” from Config menu -> Quick add -> 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 </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</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” by just performing the above 3 tasks (add the Traffic sender and SIP server accounts from the “Users and devices” form and then add the new server to the routing from 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'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><b><span style='font-size:14.0pt'>Multiple SIP servers:</span></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>You can use one SBC SIP 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'><u><span style='font-size:14.0pt'>Registrar routing:</span></u></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 SIP 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'><i><span style='line-height:115%'>If the “<span style='color:#984806'>routingforregister</span>” is set to 0 (default), then it will forward all registrations to your default SIP server which you have created during the configuration wizard.</span></i></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 might consider to set-up a separate SIP SBC 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'><u><span style='font-size:14.0pt'>Call routing:</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; 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 SIP clients) 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 are handled automatically.</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 case if the calls have to be routed to the target domain specified by the clients, then set the <span style='color:#984806'>allowupperserverselection</span> config to <span style='color:#984806'>1</span> and the <span style='color:#984806'>upperserverlookup</span> config to <span style='color:#984806'>2</span>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>In this case you should just add all your SIP servers with equal priority for the Routing and the call will be routed to the domain/IP received from the client (target URI of the SIP requests).</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="_Toc103673773"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>SIP Clients and Trunks</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'>Check “how to connect?” from the “Help” menu for the exact details about SIP clients configuration.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>For a quick test, register with two softphone 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 SBC 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="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'>username / password: loaded from your SIP server (or from the SBC if you created a special user with username/password authentication)</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'><b><span style='font-size:14.0pt'>Add SIP Client:</span></b></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 SIP 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 SIP SBC 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 SIP client will register or make call, a user entry is automatically created by the SBC. 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 SIP SBC (The users are created with “Username” based authentication only, however the SIP SBC 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 “Config” menu -> “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="_Toc103673774"></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 SIP SBC to properly route the calls to your SIP server(s).</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;line-height: normal'><b><u><span style='font-size:14.0pt'>Client driven routing</span></u></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The outbound routing can be influenced also from the client side. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>In this case the client will suggest the target upper SIP server with the INVITE target URI (so you will configure the gateway as a SIP proxy in your app and will set the SIP domain to the target upper SIP server) or by the X-Sy.Uppersrv (address), X-Sy.Uppersrvd (domain), X-Sy.Upperproxy (proxy) extra SIP headers. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>This is considered by the gateway only if <i>allowupperserverselection</i> global configuration is set to <i>1</i>. It can be set also from the Config Wizard -> “SIP Server” page -> “Allow client driven upper server selection” checkbox. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>In this case you should still create the SIP Server user entries and add them to the Routing (usually with equal priority, so the target server will be selected based on your app preferences)</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'>Technical details about the upper SIP server select algorithm in short (in priority order. Top/1 is higher priority):</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-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span>if called is SIP server:</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'>1.<span style='font:7.0pt "Times New Roman"'> </span>Sy.Upperserver or target URI sent in INVITE</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'>2.<span style='font:7.0pt "Times New Roman"'> </span>UpperServer cache (usrv = config.GetUpperServer)</p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'>3.<span style='font:7.0pt "Times New Roman"'> </span>SIP server settings from DB </p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'>4.<span style='font:7.0pt "Times New Roman"'> </span>B huser from cache (settings from last target user REGISTER)</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-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span>if called is user: </p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'>1.<span style='font:7.0pt "Times New Roman"'> </span>B huser from cache (settings from last target user REGISTER)</p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'>2.<span style='font:7.0pt "Times New Roman"'> </span>Enduser settings from DB</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="_Toc103673775"></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, 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 SIP 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="_Toc103673776"><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 SIP SBC.</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 record 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 CDRs 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 CDRs by using the “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="_Toc103673777"><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 SIP SBC 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 ->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 ->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 ->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 Files menu -> Folders -> 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 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 SIP SBC 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="_Toc103673778"><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 SBC 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> <h3 style='margin-top:0in'><a name="_Toc103673779"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Upgrade</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 SBC 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 first!) 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> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>In case if you wish to upgrade while keeping all your settings and data, then just do 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%'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Stop the service (“mserver” from windows services or from Control menu -> Stop 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%'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Backup (copy all files, except the logs, just in case if something will go wrong with the upgrade)</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%'>Install (just run the new installer, which will overwrite all binaries and will keep your old settings and data intact)</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%'>Start the service</span></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="_Toc103673780"><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 the SIP SBC 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 SBC will automatically create nightly backups and auto-delete old backups (Search for “backup” in the global configuration to change this).<a name="_Additional_Settings"></a></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="_Toc103673781"></a><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 below discussed settings have to be changed only if you have some special requirements, otherwise the default settings are optimal out of the box.</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> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>The SIP SBC has a long list of configurable settings which you might adjust to turn on/off features or to adjust the SBC 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="_Toc103673782"></a><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 rtc 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%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc103673783"></a><a name="_Toc478373690"></a><a name="_Toc478281599"><span style='font-size:14.0pt;line-height:115%;font-family: "Calibri",sans-serif'>Change SBC 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 SBC 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 SBC 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) and 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 SIP SBC 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><a name="_Toc103673784"><span style='font-size:14.0pt;line-height:115%'>Change SIP server address</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;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>This is about changing your SIP server IP address or domain name (the upper SIP server where the calls are routed and received from) if you are using a single SIP server. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>For example if you migrated your SIP server to a new location or if you have to change the upper SIP server details for any reason.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>In case if you are using a licensed SIP SBC, make sure that your license allows the usage also with the new SIP server address.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'><br> <b>GUI configuration (this will not rewrite the upper server for existing users):</b></span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>You might go trough the Configuration Wizard again and configure your new SIP server.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>Alternatively just search for “fwdregistrations_” on the “Configurations” form and rewrite it to your new server address.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>Add the new SIP server user record at the "Users and Devices" form.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>Use the Routing form and specify your custom rules for the server selection.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><b><span style='font-size:14.0pt;line-height:120%'>From SQL:</span></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>Replace OLDIP to your old SIP server IP and the NEWIP to the new SIP server IP address and then run the following SQL commands to change the upper SIP server address (copy-paste to the “Direct Query” form and hit the “Go” button):</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%;color:#984806'>update tb_users set hwid = REPLACE(hwid,'OLDIP','NEWIP'), ContractComment = REPLACE(ContractComment,'OLDIP','NEWIP'), ip = REPLACE(ip,'OLDIP','NEWIP'), authip = REPLACE(authip,'OLDIP','NEWIP'), domainname = REPLACE(domainname,'OLDIP','NEWIP'), proxyaddress = REPLACE(proxyaddress,'OLDIP','NEWIP'), transip = REPLACE(transip,'OLDIP','NEWIP'), name = REPLACE(name,'OLDIP','NEWIP'), username = REPLACE(username,'OLDIP','NEWIP')</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%;color:#984806'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%;color:#984806'>update tb_users_authip set authip = REPLACE(authip,'OLDIP','NEWIP')</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%;color:#984806'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%;color:#984806'>update tb_settings set valstr = REPLACE(valstr,'OLDIP','NEWIP')</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>If you (also) use a domain name, then replace OLDIP to your old SIP server domain and the NEWIP to the new SIP server domain name and then run the SQL statements again.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><b><span style='font-size:14.0pt;line-height:120%'>Using the SBC with multiple SIP servers:</span></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal;text-autospace:none'><span style='font-size:14.0pt'>The SIP SBC can be also used with multiple SIP servers. To be able to route the users to their respective servers (register, call, chat and other sessions) you have the following possibilities:</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;text-autospace:none'><span style='font-size:14.0pt'>1.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Specify the upper server from your SIP client instead (as the SIP server domain for your app config). <br> To allow client driven upper server selection, set the <span style='color:#632423'>allowupperserverselection </span>global config option is set to <span style='color:#632423'>true</span>.</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;text-autospace:none'><span style='font-size:14.0pt'>2.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Specify the upper server from the gateway/server routing rules:</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;line-height:normal; text-autospace:none'><span style='font-size:14.0pt'> Add the new SIP server to the “Users and Devices”</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal;text-autospace:none'><span style='font-size:14.0pt'> Use the Routing form and specify your custom rules for the server selection.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal;text-autospace:none'><span style='font-size:14.0pt'> For the routing rules to be applied also for register requests, you need to set the <span style='color:#632423'>routingforregister </span>global config option is set to <span style='color:#632423'>1</span>.</span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in;line-height: normal;text-autospace:none'><span style='font-size:14.0pt'>3.<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt'>Both:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal;text-autospace:none'><span style='font-size:14.0pt'> You can use both routing rules and upper server suggestions from your SIP clients.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal;text-autospace:none'><span style='font-size:14.0pt'> If the allowupperserverselection settings is true, then the client suggestion will overwrite your routing rules.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal;text-autospace:none'><span style='font-size:14.0pt'>For all cases, make sure to create the necessary records from the “Users and devices” form for all your SIP servers:</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;text-autospace:none'><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'>Add them as “SIP Server” (for outbound) and as “Traffic Senders” (for inbound, usually with IP authentication).</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;text-autospace:none'><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'>Then configure them in the “Routing” after your needs to specify what kind of users/calls have to be routed to which SIP server. </span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;line-height:normal; text-autospace:none'><span style='font-size:14.0pt'>For example you can just list them under the “default” pattern with equal priority for load balancing or configure any rules after your requirements.</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%'> </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="_Toc103673785"></a><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%'>The SIP SBC will listen on several ports to offer its services: SIP signaling, RTP/RTCP ports range 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%'> </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 box, then we recommend to assign a secondary IP for your server and bind the SBC 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 SIP SBC on the same box where your softswitch is running, then assign a separate IP (bindip) to the SBC if possible. If this is not possible, then make sure to change the ports on the SBC to avoid conflicts (if your SIP server is listening on 5060, then set a different “<span style='color:#984806'>localport</span>” for the SBC).</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 most important ports can be configured from the Configuration Wizard. Other ports can be configured in the global config.</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 exact port numbers used by the SBC service can be listed from MManage -> Config menu -> Network - > Active 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%'>Here is the default <b>list of ports</b> used by the SBC 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 TCP (main server port for various purposes such as API and others. Default is 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; 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)</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 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'>localport+1 TCP (for secure SIP. 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 access for voice recording . 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="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'>TCP: 9885, 9886, 9889 (optional ports for admin console and logs)</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 SBC are not used also by some other application such as a local web server (in this case either change the SBC 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> <h3 style='margin-top:0in'><a name="_Toc103673786"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>RTP Routing</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'>By default, the SBC will try to handle RTP/RTCP routing automatically by routing the RTP when the call endpoints are behind NAT and off-route when direct peer to peer RTP routing will likely succeed. </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: normal'><span style='font-size:14.0pt'>However, this default behavior can be fully reconfigured if you have any specific 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;line-height: normal'><span style='font-size:14.0pt'>RTP routing can be reconfigured either globally or by endpoint.</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'>The global (for all endpoints and call) RTP routing can be configured from Config menu -> Configurations -> RTP Relay. </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'>Set to “Bypass” to allow direct peer to peer media between the endpoints; Set to “Route” to allow the SBC to route the RTP/RTCP; Set to “Force” to disable all kind of peer to peer functionality;</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'>RTP routing by endpoint can be configure from the “Users and Devices” form -> “Functions” page -> “Media relay” settings. You can configure it for Endusers, SIP Servers and Traffic Senders.</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'>Set to “Always route RTP” to force RTP routing for all calls via that user or device; Set to “SDP correction if necessary” to disable RTP routing by the gateway; Set to other value if RTP routing should depend on caller/called NAT be</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'>RTP routing will be automatically forced if the SBC has to process the media, for example for call recording, conferencing or DTLS decoding.</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'>See the admin guide “Media Routing” FAQ point for more details about this topic.</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="_Toc103673787"></a><a name="_Toc478373692"></a><a name="_Toc478281601"><span style='font-size:14.0pt;line-height:115%;font-family: "Calibri",sans-serif'>SBC 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 SBC 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'><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 SIP SBC 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 (SIP clients or SIP server on the internet. If both of these are behind NAT, then there is no need to enable these ports). </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 SIP 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 SIP 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%'>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%'>SBC 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. (SIP 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%'>SBC, SIP server and SIP 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.</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%'>SBC and SIP 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%'>SBC behind NAT, SIP 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> <h3 style='margin-top:0in'><a name="_Toc103673788"></a><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'> </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'> </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.</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.</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="_Toc103673789"></a><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'>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="_Toc103673790"></a><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 SIP SBC 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 SIP client and SIP server capabilities.</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%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc103673791"></a><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 SBC will try to avoid transcoding when possible by negotiating a common codec between caller and called parties. If necessary, it can convert automatically between the SIP client and server by sending a re-INVITE with all supported codec’s when a 488 answer is received from the caller.</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 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%'>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%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Valid payload values are the followings:</span></p> <p class="MsoListParagraphCxSpFirst" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>PCMU (G.711 u-Law): 0</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>PCMA (G.711 A-Law): 8</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>GSM (GSM 06.10): 3</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>G.722: 9</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>G.723: 12</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>G.729: 18</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>iLBC: 97</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Speex: 104</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Speex wideband: 105</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Speex ultra wideband: 106</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Opus: 110</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Opus wideband: 111</span></p> <p class="MsoListParagraphCxSpLast" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:14.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Opus ultra wideband: 112</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%'>Example: If your SIP Server has support for G.729 and PCMU, then you should enter “18,0” for the choosecodecs user field (or in the “Users and devices” form -> “Functions” tab -> “Allowed codec” field.</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%'>Be aware that codec transcoding requires a high amount of CPU usage. For example one CPU (core) can handle around maximum 20-200 simultaneous transcodings between PCMU and G729 on full load (depending on your CPU type).</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%'>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 SIP SBC 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%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Example configuration if </span><span style='line-height:115%'>(caller A has only G.711 codec and) </span><span style='font-size:14.0pt;line-height:115%'>called B accepts only G.729:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%;color:black'>For the B user set the user convertcodecs field to 18 and the needcodecconversion field to 1 (The same can be done also from the SBC GUI -> Users and devices -> Functions tab).</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 all kind of transcoding on your server, run the following queries:</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:14.0pt; line-height:115%;color:#984806'>update tb_users set needcodecconversion = ‘’, choosecodecs=’’</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:14.0pt; line-height:115%;color:#984806'>update tb_settings set valstr = ‘false’ where keystr = ‘convertcodecsforced’</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:14.0pt; line-height:115%;color:#984806'>update tb_settings set valstr = ‘false’ where keystr = ‘enabletranscoding’</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:14.0pt; line-height:115%;color:#984806'>update tb_settings set valstr = ‘0’ where keystr = ‘fs_transcode’</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:14.0pt; line-height:115%;color:#984806'>update tb_settings set valstr = ‘0’ where keystr = ‘autocodecconvert’</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="_Toc103673792"></a><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%'>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).</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%'>Note: call recording can make the routing and media path longer, disabling peer to peer calls 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%'>You can enable/disable call recording for all users at once from MManage -> Config menu -> Configurations -> Recording -> Calls.</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='line-height:115%'>Recording can be also enabled/disabled by SQL commands:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='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='font-size:9.0pt;line-height:115%;color:#984806'>update [tb_users] set record = 0, RouteRTPCaller = 1, RouteRTPCalled = 1 where type in (0,5,9)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:9.0pt;line-height:115%;color:#984806'>ALTER TABLE [tb_users] DROP CONSTRAINT [DF_tb_users_record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:9.0pt;line-height:115%;color:#984806'>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='font-size:9.0pt;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%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='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='font-size:9.0pt;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='font-size:9.0pt;line-height:115%;color:#984806'>ALTER TABLE [tb_users] DROP CONSTRAINT [DF_tb_users_record]</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:9.0pt;line-height:115%;color:#984806'>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:9.0pt;line-height:115%;color:#984806'>update tb_settings, set valstr = ‘7’ where keystr = ‘defroutertp’</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='line-height:115%'> </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='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:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:9.0pt;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='font-size:9.0pt;line-height:115%;color:#984806'>alloweuserusercalls=0 (disable upper server routing and media path bypass)</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:9.0pt;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='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc103673793"></a><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> <h3 style='margin-top:0in'><a name="_Toc103673794"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>WebRTC</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 interested in WebRTC-SIP protocol conversion then you should use the </span><a href="https://www.mizu-voip.com/Software/WebRTCtoSIP.aspx"><span style='font-size:14.0pt;line-height:115%'>MRTC gateway</span></a><span style='font-size:14.0pt;line-height:115%'> which is actually an SBC with the WebRTC module included by default.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>However WebRTC support can be easily added also into any existing SIP SBC. </span><a href="mailto:info@mizu-voip.com?subject=WebRTC%20upgrade"><span style='font-size:14.0pt;line-height:115%'>Contact</span></a><span style='font-size:14.0pt;line-height:115%'> Mizutech to perform the upgrade.</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="_Toc103673795"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Push notifications</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 SIP SBC has support also for VoIP push notifications. This is a useful feature to improve the availability of mobile SIP clients by sending a push notification on incoming call or text message which will wake-up the client app, thus the call/chat can be delivered even if the app is closed or the device is sleeping.</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%'>Follow </span><a href="https://www.mizu-voip.com/Portals/0/Files/VOIP_Push_notifications.pdf"><span style='font-size:14.0pt;line-height:115%'>this documentation</span></a><span style='font-size:14.0pt;line-height:115%'> to enable push notification and integrate push support into your Android/iOS/Web app. </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="_Toc103673796"></a><a name="_API"></a><span style='font-size:14.0pt;line-height:115%;font-family:"Calibri",sans-serif'>API</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 gateway exposes also an API which can be used to perform or automate various tasks.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The API documentation can be found here: </span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><a href="https://www.mizu-voip.com/Portals/0/Files/VoIP_Server_API.pdf"><span style='font-size:14.0pt;line-height:115%'>https://www.mizu-voip.com/Portals/0/Files/VoIP_Server_API.pdf</span></a></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 example all upper server registrations can be removed with a request like this:</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><a href="http://sbcserveraddress.com/mvapireq/?apientry=upperunreg&u_username=all&authkey=KEY&authid=USERNAME&authmd5=MD5&authsalt=SALT&now=555"><span style='font-size:14.0pt;line-height:115%'>http://sbcserveraddress.com/mvapireq/?apientry=upperunreg&u_username=all&authkey=KEY&authid=USERNAME&authmd5=MD5&authsalt=SALT&now=555</span></a></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%'>Please note that for automation you can use also other tools such as “Scheduled Tasks” from the admin client or direct SQL </span><a href="https://www.mizu-voip.com/Portals/0/Files/VoIP_Database_Interface.pdf"><span style='font-size:14.0pt;line-height:115%'>database connection</span></a><span style='font-size:14.0pt;line-height:115%'> to alter any data such as user records (tb_users).</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%'>See </span><a href="https://www.mizu-voip.com/Portals/0/Files/VoIP%20Integration.pdf"><span style='font-size:14.0pt;line-height:115%'>this guide</span></a><span style='font-size:14.0pt;line-height:115%'> for various integration possibilities.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><a name="_First_test_calls"></a><span style='font-size:14.0pt;line-height:115%'> </span></p> <h3 style='margin-top:0in'><a name="_Toc103673797"></a><a name="_Toc478373709"></a><a name="_Toc478281616"></a><a name="_Logs"></a><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>Logs</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 find the SIP SBC logs from Files menu -> Folders -> 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 SBC (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.</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="_Toc103673798"></a><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 SBC is based on our Softswitch core which implements a long list of security measurements which are also applicable for the SIP SBC. 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="_Toc103673799"><span style='font-size:14.0pt; line-height:115%;font-family:"Calibri",sans-serif'>High availability</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;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>HA, failover and redundancy is supported for both downstream and upstream.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt;line-height: 120%'><span style='font-size:14.0pt;line-height:120%'>You can use a second backup SBC, multiple SBC’s for the same SIP server, handle multiple SIP servers by the same gateway or for the highest availability: use multiple gateways with multiple SIP servers.</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 high-availability you have a few options:</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%'>UAC failover: endpoint failover to secondary gateway if there are no or bad response of the REGISTER (or OPTIONS/INVITE requests).</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%'>This doesn't depend on any server side or networking feature and it can be handled entirely by your client app.</span></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in'><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%'>Automatic or manual IP or domain rewrite on failure</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%'>DNS SRV records</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%'>BGP or dynamic IP auto-reconfiguration on failure</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><a href="https://www.mizu-voip.com/Software/SIPLoadBalancer.aspx"><span style='font-size:14.0pt;line-height:115%'>SIP Load Balancer</span></a></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 other mechanism what you already using with your VoIP servers</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 SBC itself also has several built-in mechanism to failover among your SIP servers:</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%'>Basic failover (auto failover to other server(s) on upper server failure)</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%'>Routing automatic (auto prioritization, based on BRS 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'><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%'>Routing rules (upper server prioritizations/failover after pattern and/or destination priority)</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%'>Re-Register (redirect the registration to a working server on bad or no response)</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%'>Re-Dial (in-call failover when a bad or no response is received from the upper SIP 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%;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:14.0pt;line-height:115%'>Or simply you might use a separate SBC for each of your SIP servers</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: 120%'><span style='font-size:14.0pt;line-height:120%'>More details can be found </span><a href="https://www.mizu-voip.com/Solutions/LargescaleVoIP.aspx"><span style='font-size:14.0pt;line-height:120%'>here</span></a><span style='font-size:14.0pt;line-height:120%'> and </span><a href="https://www.mizu-voip.com/Portals/0/Files/VoIP_Server_HA_VoIP.pdf"><span style='font-size:14.0pt;line-height:120%'>here</span></a><span style='font-size:14.0pt;line-height:120%'>.</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="_Toc103673800"></a><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 SBC 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 SIP client and SIP server so they can be solved only by extra features provided from the SIP SBC or on your Softswitch (for example conference). Other features don’t depend on SIP but might 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'>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'>IM/Chat</span></b><span style='font-size:14.0pt'>: fast chat routing between SIP clients based on SIP MESSAGE </span><a href="https://www.ietf.org/rfc/rfc3428.txt"><span style='font-size:14.0pt'>RFC 3428</span></a><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%'>DTMF</span></b><span style='font-size:14.0pt;line-height:115%'>: For DTMF to work your SIP server and SIP 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 SBC 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 SIP client use this method.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><b><span style='font-size:14.0pt;line-height:115%'>Call forward</span></b><span style='font-size:14.0pt;line-height:115%'>: can be enabled from users and devices form -> functions tab. For call forward to work via your server, it needs to support SIP response code 301 (Moved Permanently) and/or 302 (Moved Temporarily).</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). 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'>RFC 3515</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'>Video</span></b><span style='font-size:14.0pt'>: Fast video stream routing between endpoints</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'><b><span style='font-size:14.0pt;line-height:115%'>Presence</span></b><span style='font-size:14.0pt;line-height:115%'>: Fast presence routing between SIP clients. For presence to work via your server your software needs support for PUBLISH/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="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 in MManage</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 SBC 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 SBC 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'><b><span style='font-size:14.0pt;line-height:115%'>DTMF triggered actions:</span></b></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>The gateway has built-in functions to be triggered by DTMF digits, useful for endpoints with no conference or transfer support. For these to work the PBX module have to be enabled and the <span style='color:#984806'>handledtmfevent</span> global config option must be set to <span style='color:#984806'>2</span>.</span></p> <p class="MsoNormal" style='margin-bottom:0in;margin-bottom:.0001pt'><span style='font-size:14.0pt;line-height:115%'>Here are the defined keys:</span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:12.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:12.0pt;line-height:115%'>create conference or add new user to conference: *1*number# </span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:12.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:12.0pt;line-height:115%'>unattended transfer: *9*number# </span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:12.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:12.0pt;line-height:115%'>transfer with consultation : *8*number# </span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:12.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:12.0pt;line-height:115%'>talk with new client while in transfer: 1</span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:12.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:12.0pt;line-height:115%'>talk with old client while in transfer: 3</span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:12.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:12.0pt;line-height:115%'>disconnecting last conference party: *5* </span></p> <p class="MsoListParagraph" style='margin-top:0in;margin-right:0in;margin-bottom: 0in;margin-left:.75in;margin-bottom:.0001pt;text-indent:-.25in'><span style='font-size:12.0pt;line-height:115%;font-family:"Courier New"'>o<span style='font:7.0pt "Times New Roman"'> </span></span><span style='font-size:12.0pt;line-height:115%'>disconnecting conference party: *6*number#</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 SBC 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 SBC also has a HTTP API which can be used for various tasks. See the </span><a href="start.htm#_API"><span style='font-size:14.0pt'>API chapter</span></a><span style='font-size:14.0pt'> and 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 SIP SBC 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%'>The SBC also does its best to compensate on missing support. For example 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> <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 SIP SBC)</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 SIP SBC)</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> <h3><a name="_Toc103673801"><span style='font-size:14.0pt;line-height:115%'>Resources</span></a></h3> <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 class="MsoHyperlink"><span style='font-size:14.0pt; font-family:Symbol;color:windowtext;text-decoration:none'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><a href="https://www.mizu-voip.com/Software/SIPSBC.aspx"><span style='font-size:14.0pt'>SIP SBC Home Page</span></a></p> <p class="MsoListParagraphCxSpMiddle" style='margin-top:0in;margin-right:0in; margin-bottom:0in;margin-left:.25in;margin-bottom:.0001pt;text-indent:-.25in; line-height:normal'><span style='font-size:14.0pt;font-family:Symbol'>·<span style='font:7.0pt "Times New Roman"'> </span></span><a href="https://www.mizu-voip.com/Portals/0/Files/SIP_SBC_Setup.exe"><span style='font-size:14.0pt'>Download</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 class="MsoHyperlink"><span style='font-size:14.0pt; font-family:Symbol;color:windowtext;text-decoration:none'>·<span style='font:7.0pt "Times New Roman"'> </span></span></span><a href="mailto:info@mizu-voip.com?subject=SIP%20SBC"><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
Quick setup
Features
Requirements
Install
Configuration Wizard
Troubleshooting
MizuManage
User Management
SIP Servers
SIP Clients and Trunks
Outbound Routing
Inbound Routing
CDR
Statistics and Monitoring
Start/Stop
Upgrade
Backup/Restore
Settings
Global Configuration
Change SBC IP address
Change SIP server address
Ports
RTP Routing
SBC behind NAT
Calls between endusers/extensions
Multiple registrations and call fork
Codec
Transcoding
Call recording
Chat recording
WebRTC
Push notifications
API
Logs
Security
High availability
Extra features
Resources
About
API
Backup/Restore
Call recording
Calls between endusers/extensions
CDR
Change SIP server address
Change SBC IP address
Chat recording
Codec
Configuration Wizard
Extra features
Features
Global Configuration
High availability
Inbound Routing
Install
Logs
MizuManage
Multiple registrations and call fork
Outbound Routing
Ports
Push notifications
Quick setup
Requirements
Resources
RTP Routing
SBC behind NAT
Security
Settings
SIP Servers
SIP Clients and Trunks
Start/Stop
Statistics and Monitoring
Transcoding
Troubleshooting
Upgrade
User Management
WebRTC
SIP SBC
