[Freeswitch-users] RTP to SRTP Freeswitch 1.4

[Volodymyr Fedorov]

Hello, maybe transfer makes something bad with your call. Try this before
GATEWAY in dialplan:
<action application="set" data="proxy_media=false"/>
<action application="set" data="bypass_media=false"/>
<action application="export" data="zrtp_secure_media=false"/>
<action application="export" data="nolocal:rtp_secure_media=true"/>


On Tue, Aug 25, 2015 at 5:09 AM, Gonzalo Gasca Meza <gascagonzalo at gmail.com>
wrote:

> I have the following scenario:
>
> SIP Phone (RTP/SIP UDP) ---> Freeswitch ---> SRTP/TLS --> ITSP
>
> Currently when I place a call Freeswitch is not sending SRTP SDES in SDP.
> It does the TLS connection but SDP contains only AVP codecs.
> I have used the rtp_secure_media set to true and also export but crypto
> caps are not attached. I can see them listed, but nothing happens.
> Logs:
>
> http://pastebin.com/DgZ1Lejp
>
> sofia_status
> external::itsp-tls gateway   sip:gonzalo at tls.itsp.com:5061:5061 NOREG
>
> dialplan/public.xml
>     <extension name="public_itsp_tls">
>       <condition field="destination_number" expression="^(80\d{11})$">
>           <action application="set" data="rtp_secure_media=true"/>
>           <action application="export" data="rtp_secure_media=true"/>
>           <action application="log" data="INFO Processing call to itsp"/>
>           <action application="transfer" data="$1 XML default"/>
>       </condition>
>     </extension>
>
> dialplan/default.xml
> <extension name="US_ITSP_Secure">
>         <action application="log" data="INFO Processing REGEX"/>
>         <condition field="destination_number" expression="^80(\d{11})$">
>                 <action application="set" data="dialed_number=+$1"/>
>                 <action application="log" data="INFO Processing US call to
> ITSP Secure ${dialed_number}"/>
>                 <action application="bridge"
> data="sofia/gateway/itsp-tls/${dialed_number}"/>
>          </condition>
>     </extension>
>
> sip_profiles/external/sipsitspTrunk.xml
> <include>
> <gateway name="itsp-tls">
> <param name="tls-version" value="tlsv1"/>
> <param name="register-transport" value="tls"/>
>   <param name="contact-params" value="transport=tls"/>
> <param name="realm" value="itsp-tls"/>
> <param name="proxy" value="tls.itsp.com:5061"/> <!-- (your configured
> Trunk SIP URI) -->
> <param name="register" value="false"/>
> </gateway>
> </include>
>
>
>
>
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-- 
Best regards,
Volodymyr
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