Hi Zaka, I know Juha already answered some of your questions, but I simply wanted to add some additional information regarding your third question.
Yes disabling would do the job, but if you plan to let your customers use Chrome here is some info I ran into using SIP over WebSocket with WebRTC-Clients (most likely you know this already): - Chrome does not allow to use WebRTC on unsecure channels. As a result if you switch to SSL (for your WebRTC-Client Page) Chrome is blocking normal Websocket communication so then I had to turn on WSS in Kamailio which included a TLS setup. Best Regards Dimitry Nagorny Trainee Von: sr-users [mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Zaka Gesendet: Montag, 4. Juli 2016 15:07 An: Kamailio (SER) - Users Mailing List <sr-users@lists.sip-router.org> Betreff: [SR-Users] Kamailio WEBRTC questions/ confusions Dear List: I wonder if this has been asked before. Please ignore or point to the appropriate source, if needed. Simply, I have little experience with SIP servlets implementing WEBRTC/ Click2Call etc... Considering a Call Centre scenario offering a WEBRTC solution. Following are the confusions regarding requirements: 1: A Kamailio Box (running WEBSOCKET.config) 2: Customer will use web browser (websocket compatible of course) and enter address of the KAMAILIO Box or we shall need another KAMAILIO to statelessly forward the call to next available operator using Hunt Group or similar scheme? 3: If we don't plan to implement TLS/ MSRP (disabling these in the config file should be suffice?) 4: Last, but not the least, how will the users send request for login from browser? (assuming they have accounts created in the DB on Kamailio Box running WEBSOCKET supported KAMAILIO) With anticipatory thanks & regards, Zaka
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