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
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" idokan@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
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anfecora -
ik