You didn't mention what rev you're using, If you're in 1.4 then its rtp_secure_media, please see latest configs and extensive docs in vars.xml about srtp and all the nice knobs you can use to tweak it. On Tue, Sep 9, 2014 at 12:52 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP.
I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP). But still FreeSwitch does not use RTP/SAVP for leg-B.
Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.
<!-- Since we have inbound-late-negotation on by default now the above behavior isn't the same so you have to do one extra step. -->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/> <!-- Offer SRTP on outbound legs if we have it on inbound. --> <action application="export" data="sip_secure_media=true"/> </condition>
By commenting/un-commenting "sip_secure_media=true" above, it did not worked.
Any help with this would be greatly appreciated.
-- Thanks CWM
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