Does it work on the latest 1.4 release with the config change he mentioned? On Sep 10, 2014, at 1:48 AM, Chandrakant Marathe <cwmarathe@gmail.com> wrote:
Sorry Brian, I should have told the version. My bad. I am using 1.2 stable release. When I hit "git branch -av", it outputs -
* v1.2.stable 2b62885 fs_cli: fix compiler error on CentOS 6 caused by recent short uuid logging change
And from FreeSwitch console, when I hit "version" command, it shows -
FreeSWITCH Version 1.2.24+git~20140630T213113Z~2b62885f21~32bit (git 2b62885 2014-06-30 21:31:13Z 32bit)
I went through "vars.xml" and found only one config related with "rtp_secure_media" and that to related with zrtp
<X-PRE-PROCESS cmd="set" data="zrtp_secure_media=true"/>
Is there any more configuration to do in v1.2 so as to enable end-to-end secure call or shall I move to v1.4 beta?
Thanks. -- CWM
On Tue, Sep 9, 2014 at 9:04 PM, Brian West <brian@freeswitch.org> wrote: 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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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org