<div dir="ltr"><div><div><div><div><div><div>Thank you Carlos, I was taking a look of you work and it is really interesting. I was trying to somehow apply the conversion that you did with Websockets on my conversion/encryption but I haven't reached into the solution.<br>
<br></div>I was able to install Kamilio with the rtpproxy-ng module and rtpengine as media relay. Also I was able to do the TLS encryption.<br><br></div>But when I set the flags for rtpengine, I can just set the whole media as SRTP or RTP. I am not able to do one side in SRTP and the other with RTP. (I wanted Kamailio with rtpengine to encrypt calls).<br>
<br></div>Is that conversion able with some configuration of flags? Which documentation could help me to understand how to do this process?<br><br></div>If someone has any clue... Thank you in advance.<br><br></div>Regards,<br>
</div>Daniel.<br><div><div><div> <br></div></div></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-06-26 15:59 GMT+02:00 Carlos Ruiz Díaz <span dir="ltr"><<a href="mailto:carlos.ruizdiaz@gmail.com" target="_blank">carlos.ruizdiaz@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi,<div><br></div><div>you are on the right track. Mediaproxy-ng/rtpengine does the conversion of SDP profiles for you, so basically, you will only need to flag the call with the right parameters and rtpengine will do the rest. </div>
<div><br></div><div>Check out the webrtc example that comes with Kamailio, or my example [1]. You can use this as a starting point to understand how rptengine does the translations.</div><div><br></div><div>And in regards of TLS, check out the tls module documentation, but in summary, you can choose to encrypt communication in one socket, and maintain clear UDP/TCP in another. Kamailio will handle the routing among the two.</div>
<div><br></div><div>[1] <a href="https://github.com/caruizdiaz/kamailio-ws" target="_blank">https://github.com/caruizdiaz/kamailio-ws</a></div><div><br></div><div>Regards,</div><div>Carlos</div><div><br></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote"><div><div class="h5">On Thu, Jun 26, 2014 at 3:35 AM, Dani Kamailio <span dir="ltr"><<a href="mailto:dani.kamailio@gmail.com" target="_blank">dani.kamailio@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">
<div dir="ltr"><div>Hello,<br><br>I need to build a VoIP system who receives SIP and RTP
traffic in a public IP and encrypt both of them with TLS and SRTP respectively. The
main point is to have security inside of the local network (I know it may sound unuseful).<br><br>So, I
was trying to build the whole system in Kamailio but I got stucked with
the RTP to SRTP bridge and I do not really know how to do it. I know
that there are some modules like rtpproxy-ng and rtpengine as media
relay that can handle SRTP... any ideas?<br></div><div>I just want to make sure that I am in the right way.<br></div><div><br></div><div>If it could be done in Kamailio, have I to write the code to encrypt RTP, like the algorithm or something?</div>
<div><br></div>Would it be convenient to send the RTP packets to Asterisk? (and Asterisk would somehow encrypt them?)<br>
<div><br>I am pretty new with all of this, any help will be apreciate,<br>thank you.</div></div>
<br></div></div>_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br></blockquote></div><span class="HOEnZb"><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div dir="ltr">Carlos<div><a href="http://caruizdiaz.com" target="_blank">http://caruizdiaz.com</a></div><div><a href="http://ngvoice.com" target="_blank">http://ngvoice.com</a></div>
<div><a href="tel:%2B52%2055%203048%203303" value="+525530483303" target="_blank">+52 55 3048 3303</a></div></div>
</font></span></div>
<br>_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br></blockquote></div><br></div>