Thanks to all who have replied - the problem has been found (my fault...). The variables set above (with set/export) work as intended. I simply had a clause in the dial string that sets SRTP only if TLS is active as well. __Yehavi: 2012/11/21 Brian West <brian@freeswitch.org>
you make sure sip_secure_media=true is set inside {} on the originate or export the variable. -- Brian West brian@freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 ISN: 410*543
On Nov 21, 2012, at 7:08 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hi,
I am trying to use SRTP without TLS. The initiating side asks and get SRTP (two way) while the other side is not offered SRTP at all.
I've noted that ${sip_has_crypto} is null. I also tried setting unconditionaly sip_secure_rtp (both set and export), but it doesn't help.
Any idea what I am doing wrong?
BTW, I am using today's mornning GIT.
Thanks, __Yehavi:
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