hi, i have 2 profiles where one is purely SIP/RTP and the second is TLS/SRTP only, i have clients registering to both. If a TLS registered client calls a SIP based one everything works fine as FS uses SRTP for orginal leg and RTP for the outbound leg but if a SIP registered client calls a TLS one then the audio fails as FS tries to send RTP rather than SRTP. I have late negotiation and inherit codecs enabled. Is there any way to check in dialplan if the registered user is TLS based and use SRTP for that leg of the call and use normal SIP/RTP for the originating leg or some profile param which can force SRTP for all clients registered to the TLS profile and RTP for all clients registered to the SIP profile the way im currently getting around this is using the below but it isnt elegant <extension name="local_user_calling_check tls" continue="true"> <condition field="${sofia_contact(tls/${destination_number}@${domain})}" expression="^error\/user_not_registered$"> <anti-action application="export" data="nolocal:rtp_secure_media=true"/> </condition> </extension> then i bridge the call, if the above fails it means its a SIP to SIP call -- Regards, Bipin