FS latest FusionPBX ===================== Hi list, I read a lot from the wiki, but I'm a bit lost about securing calls. From what I read elsewhere, ssl+srtp seems to be the best solution, however the wiki only talks about tls+srtp. I modified conf/vars.xml as of the wiki & enabled xxx_ssl_enable, but put 'sslv23' instead of 'tls'; I also modified conf/directory/default.xml (from 'tls' to 'sslv23') as: <param name="dial-string" value="{sip_secure_media=${regex(${sofia_contact(${dialed_user}@${dialed_domain})}|transport=sslv23)},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}" /> is this right & sufficient? If not, what do I miss or what would be the best solution to make sure all calls will be secured both signaling & conversations? At this time, I make tests with the jitsi softphone (ex sip-commuâ€nicator) which seems to use SRTP once the SAS has been accepted from each side: that's what wireshark shows, but it also show that SIP is use instead of SIPS, is it because I move 'tls' to 'sslv23'? JY --