don't us export, set it inside {}, or on use set on a-leg. /b On Tue, Jul 31, 2018 at 9:23 AM, Chhorm Chhatra <ch.chhatra@gmail.com> wrote:
Hello,
Currently, I faced a problem regarding SRTP outbound call to user (Leg B).
The scenario is like this,
- We set up our own root CA to an IP address (e.g 192.168.0.13) - We create a server certificate for freeswitch at 192.168.0.13 - Linphone is used as SIP client and is configured to trust our root CA by default. - Linphone A is configured to register to Freeswitch vis TLS + SRTP. (One leg call to server has both SIPs and SRTP – completely secure) - Linphone B is registered to Freeswitch via TLS + SRTP, and waiting for Linphone A to call to.
(One leg call to server, e.g. 9196 (echo test), is completely secure with SRTP + SIPs)
- Unfortunately, if A call to B, only A leg has SIPs + SRTP, but Leg B is not encrypted with SRTP and SIPs at all. This causes *SRTP unprotect failed with code 7 (auth check failed)**.*
+ Dialplan Configuration
<action application="set" data="rtp_secure_media=true"/>
<action application="export" data="rtp_secure_media=true"/>
The dial-string is <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
+ Directory Configruation:
<param name="dial-string" value="{rtp_secure_media=${reg ex(${sofia_contact(${dialed_user}@${dialed_domain})}| transport=tls)},presence_id=${dialed_user}@${dialed_domain}} ${sofia_contact(${dialed_user}@${dialed_domain})}" />
My question is that, is there any configuration left that I have to set up in order to let freeswitch initiate an outbound call to Leg B correctly with SRTP and SIPs (tls)?
Any help would be really appreciated. Thank you so much. Best Regard,
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