[Freeswitch-users] SIP/RTP to TLS/SRTP calls

[Bipin Patel]

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