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<div><font color="#000000" size="2" face="Arial">Dear Kamailio users</font><font color="#000000" size="2" face="Arial">,</font></div>
<div>Opensips 1.4.0 has been installed in my linux server with TLS enabled.</div>
<div><font size="2" face="Arial">I am confusing with the scenerio of how to use SIPS/TLS to make security VoIP by Opensips.</font></div>
<div><font size="2" face="Arial">I got the information like following from console after run opensips 1.4.0 with TLS enabled.</font></div>
<div><font size="2" face="Arial"></font> </div>
<div>Listening on<br> udp: 192.168.1.11 [192.168.1.11]:5060<br> tcp: 192.168.1.11 [192.168.1.11]:5060<br> tls: 192.168.1.11 [192.168.1.11]:5061<br>Aliases:<br> tls: wavehost:5061<br>
tcp: wavehost:5060<br> udp: wavehost:5060</div>
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<div><font size="2" face="Arial">As RFC3261 said, 5061 is default port for SIP. So, there are two ports here after TLS enabled. My question is how to configure my UA to use the SIP server?</font></div>
<div><font size="2" face="Arial">In the other word, how to use SIP server with TLS enabled? My UA is SJphone 1.6 and XLite 3.0, seems they doesn't support SIPs/TLS. Is there any open source SIP UA which support TLS availiable ?</font></div>
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<div><font size="2" face="Arial">Best regards,</font></div>
<div><font size="2" face="Arial">Steven Wu</font></div></div>
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