You might need to also add asterisk 12 b2b in order to convert to simple sip to solve issues with ice on the same box. On Apr 1, 2014 11:52 AM, "ik" <ido...@gmail.com> wrote:
> Hello, > > I'm a newbie with Kamailio, and I require to connect webrtc (websockets) > based phones, into a VoIP PBX that does not support websockets. > > I wish to create/use Kamailio rules that will translate UDP to websockets > and vice versa. > > I have found few examples over the internet, but as it seems (to me), they > are just doing normal SIP operations under websockets (registration, > routing, voicemails etc). > > Is there a way to make Kamailio a broker that understand both transports, > and translate them ? > > If so, can you please point me to a documentation/example that does it > that might help me better understand it ? > > Please note that I do not have any experience with Kamailio, and just > getting started with it. > > > Thank you very much, > > Ido > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
