26 Jun
2014
26 Jun
'14
11:35 a.m.
Hello,
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).
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?
I just want to make sure that I am in the right way.
If it could be done in Kamailio, have I to write the code to encrypt RTP,
like the algorithm or something?
Would it be convenient to send the RTP packets to Asterisk? (and Asterisk
would somehow encrypt them?)
I am pretty new with all of this, any help will be apreciate,
thank you.
26 Jun
26 Jun
4:59 p.m.
Hi,
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.
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.
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.
[1] https://github.com/caruizdiaz/kamailio-ws
Regards,
Carlos
On Thu, Jun 26, 2014 at 3:35 AM, Dani Kamailio <dani.kamailio(a)gmail.com>
wrote:
Hello,
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).
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?
I just want to make sure that I am in the right way.
If it could be done in Kamailio, have I to write the code to encrypt RTP,
like the algorithm or something?
Would it be convenient to send the RTP packets to Asterisk? (and Asterisk
would somehow encrypt them?)
I am pretty new with all of this, any help will be apreciate,
thank you.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Carlos
http://caruizdiaz.com
http://ngvoice.com
+52 55 3048 3303
4 Jul
4 Jul
5:18 p.m.
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.
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.
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).
Is that conversion able with some configuration of flags? Which
documentation could help me to understand how to do this process?
If someone has any clue... Thank you in advance.
Regards,
Daniel.
2014-06-26 15:59 GMT+02:00 Carlos Ruiz Díaz <carlos.ruizdiaz(a)gmail.com>om>:
Hi,
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.
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.
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.
[1] https://github.com/caruizdiaz/kamailio-ws
Regards,
Carlos
On Thu, Jun 26, 2014 at 3:35 AM, Dani Kamailio <dani.kamailio(a)gmail.com>
wrote:
Hello,
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).
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?
I just want to make sure that I am in the right way.
If it could be done in Kamailio, have I to write the code to encrypt RTP,
like the algorithm or something?
Would it be convenient to send the RTP packets to Asterisk? (and Asterisk
would somehow encrypt them?)
I am pretty new with all of this, any help will be apreciate,
thank you.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Carlos
http://caruizdiaz.com
http://ngvoice.com
+52 55 3048 3303
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
3794
days inactive
3802
days old
2 comments
2 participants
participants (2)
-
Carlos Ruiz Díaz -
Dani Kamailio