<div dir="ltr"><div><div><div>Dear List & Dimitry:<br><br></div>Please see the following & advise:<br><br></div>1: I have not been able to make Kamailio Listen on WEBTC port (in my case 8000)<br><br></div>Please see the websocket.cfg DEFs section (with TLS/MSRP disabled)<br><br><br>#!substdef "!DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g"<br>#!substdef "!MY_IP_ADDR!10.42.0.1!g"<br>#!substdef "!MY_DOMAIN!<a href="http://callcntr.com.al">callcntr.com.al</a>!g"<br>#!substdef "!MY_WS_PORT!8000!g"<br>#!substdef "!MY_WSS_PORT!443!g"<br>#!substdef "!MY_MSRP_PORT!9000!g"<br>#!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g"<br>##!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g"<br>##!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g"<br>##!substdef "!MSRP_MIN_EXPIRES!1800!g"<br>##!substdef "!MSRP_MAX_EXPIRES!3600!g"<br><br>##!define LOCAL_TEST_RUN<br>##!define WITH_TLS<br>#!define WITH_WEBSOCKETS<br>##!define WITH_MSRP<br><br><div><div>and the output of command <br><br><b>root@callcntr:/usr/local/etc/kamailio# kamailio -Ee -l 10.42.0.1 -dd -cf websocket.cfg<br><br>0(10634) INFO: <core> [main.c:1911]: main(): private (per process) memory: 8388608 bytes<br> 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !DBURL!mysql://kamailio:kamailiorw@localhost/kamailio!g<br> 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_IP_ADDR!10.42.0.1!g<br> 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_DOMAIN!<a href="http://callcntr.com.al">callcntr.com.al</a>!g<br> 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_PORT!8000!g<br> 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WSS_PORT!443!g<br> 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_MSRP_PORT!9000!g<br> 0(10634) INFO: <core> [ppcfg.c:82]: pp_subst_add(): ### added subst expression: !MY_WS_ADDR!tcp:10.42.0.1:8000!g<br>loading modules under config path: /usr/local/lib64/kamailio/modules/<br> 0(10634) INFO: <core> [sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module<br>Listening on <br> udp: <a href="http://10.42.0.1:5060">10.42.0.1:5060</a><br> tcp: <a href="http://10.42.0.1:5060">10.42.0.1:5060</a><br>Aliases: <br> tcp: <a href="http://callcntr.com.al:5060">callcntr.com.al:5060</a><br> udp: <a href="http://callcntr.com.al:5060">callcntr.com.al:5060</a><br><br></b></div><div>Besides, when it is run without -c flag, it periodically outputs WS_CONN status with [null] members.<br><br></div><div>netstat confirms that it is only listening on port 5060. <br><br></div><div>What I am missing?<br><br></div><div>KR,<br><br></div><div>Zaka <br></div><div><br><br><b></b></div><div><b><br></b></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 5 July 2016 at 08:49, Nagorny, Dimitry <span dir="ltr"><<a href="mailto:dimitry.nagorny@robot5.de" target="_blank">dimitry.nagorny@robot5.de</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Hi Zaka,<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">I know Juha already answered some of your questions, but I simply wanted to add some additional information regarding your third question.
<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Yes disabling would do the job, but if you plan to let your customers use Chrome here is some info I ran into using SIP over WebSocket with WebRTC-Clients
(most likely you know this already):<u></u><u></u></span></p>
<p><u></u><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><span>-<span style="font:7.0pt "Times New Roman"">
</span></span></span><u></u><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Chrome does not allow to use WebRTC on unsecure channels. As a result if you switch to SSL (for your WebRTC-Client Page) Chrome is blocking
normal Websocket communication so then I had to turn on WSS in Kamailio which included a TLS setup.<u></u><u></u></span></p>
<p><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Best Regards<u></u><u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Dimitry Nagorny<u></u><u></u></span></b></p>
<p class="MsoNormal"><span style="font-size:8.0pt;font-family:"Calibri",sans-serif">Trainee</span><span style="font-size:8.0pt;font-family:"Calibri",sans-serif;color:#7f7f7f" lang="EN-US"><u></u><u></u></span></p>
<p class="MsoNormal"><span><u></u> <u></u></span></p>
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<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">Von:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"> sr-users [mailto:<a href="mailto:sr-users-bounces@lists.sip-router.org" target="_blank">sr-users-bounces@lists.sip-router.org</a>]
<b>Im Auftrag von </b>Zaka<br>
<b>Gesendet:</b> Montag, 4. Juli 2016 15:07<br>
<b>An:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>><br>
<b>Betreff:</b> [SR-Users] Kamailio WEBRTC questions/ confusions<u></u><u></u></span></p>
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<p class="MsoNormal"><u></u> <u></u></p>
<p>Dear List:<u></u><u></u></p>
<p>I wonder if this has been asked before. Please ignore or point to the appropriate source, if needed.<u></u><u></u></p>
<p>Simply, I have little experience with SIP servlets implementing WEBRTC/ Click2Call etc...<u></u><u></u></p>
<p>Considering a Call Centre scenario offering a WEBRTC solution. Following are the confusions regarding requirements:<u></u><u></u></p>
<p>1: A Kamailio Box (running WEBSOCKET.config)<u></u><u></u></p>
<p>2: Customer will use web browser (websocket compatible of course) and enter address of the KAMAILIO Box or we shall need another KAMAILIO to statelessly forward the call to next available operator using Hunt Group or similar scheme?<u></u><u></u></p>
<p>3: If we don't plan to implement TLS/ MSRP (disabling these in the config file should be suffice?)<u></u><u></u></p>
<p>4: Last, but not the least, how will the users send request for login from browser? (assuming they have accounts created in the DB on Kamailio Box running WEBSOCKET supported KAMAILIO)<u></u><u></u></p>
<p>With anticipatory thanks & regards,<u></u><u></u></p>
<p>Zaka<u></u><u></u></p>
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<br></blockquote></div><br><br clear="all"><br>-- <br><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><span style="color:rgb(51,0,51)">Best regards,</span><br style="color:rgb(51,0,51)"><br style="color:rgb(51,0,51)"><span style="color:rgb(51,0,51)">Zaka Bhatti</span><br style="color:rgb(51,0,51)"><span style="color:rgb(51,0,51)">Tirana,</span><br style="color:rgb(51,0,51)"><span style="color:rgb(51,0,51)">Republic of Albania</span><br style="color:rgb(51,0,51)"><span style="color:rgb(51,0,51)">+355 672 000 771</span><br><br><span><a href="https://al.linkedin.com/in/kamailian" target="_blank">https://al.linkedin.com/in/kamailian</a><br><br><br></span></div><div><span>C2C (Click to Call)<br></span></div><div dir="ltr"><br><a href="http://click2dial.org/u/emFrYS5iaGF0dGlAZ21haWwuY29t" target="_blank">http://click2dial.org/u/emFrYS5iaGF0dGlAZ21haWwuY29t</a><br><br><span style="color:rgb(0,0,153)">One accurate measurement is worth more than a thousand expert opinions!</span></div></div></div></div></div></div>
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